void addBinocularsSubstream0(void*)
{
binocularsStream = new FrameStreamState;
binocularsStream->content = binoculars->getContent();
Port rtpPort(BINOCULARS_RTP_PORT_NUM);
Groupsock* rtpGroupsock = new Groupsock(*env, destinationAddress, rtpPort, TTL);
//rtpGroupsock->multicastSendOnly(); // we're a SSM source
// Create a 'H264 Video RTP' sink from the RTP 'groupsock':
binocularsStream->sink = H264VideoRTPSink::createNew(*env, rtpGroupsock, 96);
ServerMediaSubsession* subsession = PassiveServerMediaSubsession::createNew(*binocularsStream->sink);
binocularsSMS->addSubsession(subsession);
binocularsStream->source = H264VideoStreamDiscreteFramer::createNew(*env,
RawPixelSource::createNew(*env,
binocularsStream->content,
avgBitRate));
binocularsStream->sink->startPlaying(*binocularsStream->source, NULL, NULL);
std::cout << "Streaming binoculars ..." << std::endl;
announceStream(rtspServer, binocularsSMS, binocularsStreamName);
}
int _tmain(int argc, TCHAR* argv[])
{
try{
char* str = "1234";
fprintf(stderr, "%s", str);
OutPacketBuffer::maxSize = 200000;
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
*env<<"mack is a good boy";
string s = "0123456789";
s = s.substr(8);
::CoInitialize(NULL);
CDshowCapInfoMgr* mgr = new CDshowCapInfoMgr();
mgr->enumAllCapInfo();
mgr->printCapDetail();
//CDShowCapInfo* pInfo = mgr->getVideoInfo(0);
//CDeviceCapture* device = new CDeviceCapture(DEVICE_CAP_VIDEO_TYPE,pInfo->getFriendlyName(),pInfo->getMediaOption(0));
CDShowCapInfo* pInfo = mgr->getAudioInfo(0);
CDeviceCapture* device = new CDeviceCapture(DEVICE_CAP_AUDIO_TYPE,pInfo->getFriendlyName(),pInfo->getMediaOption(8));
device->startCap();
UserAuthenticationDatabase* authDB = NULL;
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
char const* descriptionString
= "Session streamed by \"testOnDemandRTSPServer\"";
{
device->startCap();
char const* streamName = "h264ESVideoTest";
char const* inputFileName = "test.264";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(H264DshowCapMediaServerSubsession
::createNew(*env, inputFileName,device));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
env->taskScheduler().doEventLoop(); // does not return
while (true)
{
Sleep(1000);
}
}
catch(std::exception e)
{
printf(e.what());
}
return 0;
}
//**************************************************************************************
void StreamAddMpegFile(char* streamName, wchar_t* fileName, int channelType)
{
try
{
LogDebug(L"Stream server: add mpeg-2 ts stream %S filename:%s", streamName,fileName);
ServerMediaSession* sms= ServerMediaSession::createNew(*m_env, streamName, streamName,STREAM_DESCRIPTION,false);
sms->addSubsession(TsMPEG2TransportFileServerMediaSubsession::createNew(*m_env, fileName, false,false,channelType));
m_rtspServer->addServerMediaSession(sms);
announceStream(m_rtspServer, sms, streamName, fileName);
}
catch(...)
{
LogDebug(L"Stream server: unable to add stream %S filename:%s", streamName,fileName);
}
}
void addFaceSubstreams0(void*)
{
int portCounter = 0;
for (int j = 0; j < cubemap->getEyesCount(); j++)
{
Cubemap* eye = cubemap->getEye(j);
for (int i = 0; i < eye->getFacesCount(); i++)
{
faceStreams.push_back(FrameStreamState());
FrameStreamState* state = &faceStreams.back();
state->content = eye->getFace(i)->getContent();
Port rtpPort(FACE0_RTP_PORT_NUM + portCounter);
portCounter += 2;
Groupsock* rtpGroupsock = new Groupsock(*env, destinationAddress, rtpPort, TTL);
//rtpGroupsock->multicastSendOnly(); // we're a SSM source
setReceiveBufferTo(*env, rtpGroupsock->socketNum(), bufferSize);
// Create a 'H264 Video RTP' sink from the RTP 'groupsock':
state->sink = H264VideoRTPSink::createNew(*env, rtpGroupsock, 96);
ServerMediaSubsession* subsession = PassiveServerMediaSubsession::createNew(*state->sink);
cubemapSMS->addSubsession(subsession);
RawPixelSource* source = RawPixelSource::createNew(*env,
state->content,
avgBitRate);
source->setOnSentNALU (boost::bind(&onSentNALU, _1, _2, _3, j, i));
source->setOnEncodedFrame(boost::bind(&onEncodedFrame, _1, j, i));
state->source = H264VideoStreamDiscreteFramer::createNew(*env,
source);
state->sink->startPlaying(*state->source, NULL, NULL);
std::cout << "Streaming face " << i << " (" << ((j == 0) ? "left" : "right") << ") on port " << ntohs(rtpPort.num()) << " ..." << std::endl;
}
}
announceStream(rtspServer, cubemapSMS, cubemapStreamName);
}
void LiveRtspServer::addRtspMediaSession(const Channel& channel)
{
const std::string sSessionName = channel.ChannelName;
// Next, check whether we already have an RTSP "ServerMediaSession" for this media stream:
ServerMediaSession* sms = RTSPServer::lookupServerMediaSession(sSessionName.c_str());
Boolean bSmsExists = sms != NULL;
if (!bSmsExists)
{
VLOG(2) << "Creating Session " << sSessionName << " on RTSP server";
// Create a new "ServerMediaSession" object for streaming from the named file.
sms = createNewSMS(envir(), *this, channel, m_pFactory, m_pGlobalRateControl);
VLOG(2) << "Adding ServerMediaSession " << sSessionName;
addServerMediaSession(sms);
announceStream(this, sms, sSessionName.c_str());
}
else
{
LOG(WARNING) << "Session " << sSessionName << " already exists on RTSP server";
}
}
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
UserAuthenticationDatabase* authDB = NULL;
#ifdef ACCESS_CONTROL
// To implement client access control to the RTSP server, do the following:
authDB = new UserAuthenticationDatabase;
authDB->addUserRecord("username1", "password1"); // replace these with real strings
// Repeat the above with each <username>, <password> that you wish to allow
// access to the server.
#endif
// Create the RTSP server:
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
char const* descriptionString
= "Session streamed by \"testOnDemandRTSPServer\"";
// Set up each of the possible streams that can be served by the
// RTSP server. Each such stream is implemented using a
// "ServerMediaSession" object, plus one or more
// "ServerMediaSubsession" objects for each audio/video substream.
// A MPEG-4 video elementary stream:
{
char const* streamName = "mpeg4ESVideoTest";
char const* inputFileName = "test.m4e";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(MPEG4VideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A H.264 video elementary stream:
{
char const* streamName = "h264ESVideoTest";
char const* inputFileName = "test.264";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(H264VideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A H.265 video elementary stream:
{
char const* streamName = "h265ESVideoTest";
char const* inputFileName = "test.265";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(H265VideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A MPEG-1 or 2 audio+video program stream:
{
char const* streamName = "mpeg1or2AudioVideoTest";
char const* inputFileName = "test.mpg";
// NOTE: This *must* be a Program Stream; not an Elementary Stream
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
MPEG1or2FileServerDemux* demux
= MPEG1or2FileServerDemux::createNew(*env, inputFileName, reuseFirstSource);
sms->addSubsession(demux->newVideoServerMediaSubsession(iFramesOnly));
sms->addSubsession(demux->newAudioServerMediaSubsession());
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A MPEG-1 or 2 video elementary stream:
{
char const* streamName = "mpeg1or2ESVideoTest";
char const* inputFileName = "testv.mpg";
// NOTE: This *must* be a Video Elementary Stream; not a Program Stream
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(MPEG1or2VideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource, iFramesOnly));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A MP3 audio stream (actually, any MPEG-1 or 2 audio file will work):
// To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
//#define STREAM_USING_ADUS 1
// To also reorder ADUs before streaming, uncomment the following:
//#define INTERLEAVE_ADUS 1
// (For more information about ADUs and interleaving,
// see <http://www.live555.com/rtp-mp3/>)
{
char const* streamName = "mp3AudioTest";
char const* inputFileName = "test.mp3";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
Boolean useADUs = False;
Interleaving* interleaving = NULL;
#ifdef STREAM_USING_ADUS
useADUs = True;
#ifdef INTERLEAVE_ADUS
unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...
unsigned const interleaveCycleSize
= (sizeof interleaveCycle)/(sizeof (unsigned char));
interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
#endif
#endif
sms->addSubsession(MP3AudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource,
useADUs, interleaving));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A WAV audio stream:
{
char const* streamName = "wavAudioTest";
char const* inputFileName = "test.wav";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
// To convert 16-bit PCM data to 8-bit u-law, prior to streaming,
// change the following to True:
Boolean convertToULaw = False;
sms->addSubsession(WAVAudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource, convertToULaw));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// An AMR audio stream:
{
char const* streamName = "amrAudioTest";
char const* inputFileName = "test.amr";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(AMRAudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A 'VOB' file (e.g., from an unencrypted DVD):
{
char const* streamName = "vobTest";
char const* inputFileName = "test.vob";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
// Note: VOB files are MPEG-2 Program Stream files, but using AC-3 audio
MPEG1or2FileServerDemux* demux
= MPEG1or2FileServerDemux::createNew(*env, inputFileName, reuseFirstSource);
sms->addSubsession(demux->newVideoServerMediaSubsession(iFramesOnly));
sms->addSubsession(demux->newAC3AudioServerMediaSubsession());
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A MPEG-2 Transport Stream:
{
char const* streamName = "mpeg2TransportStreamTest";
char const* inputFileName = "test.ts";
char const* indexFileName = "test.tsx";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(MPEG2TransportFileServerMediaSubsession
::createNew(*env, inputFileName, indexFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// An AAC audio stream (ADTS-format file):
{
char const* streamName = "aacAudioTest";
char const* inputFileName = "test.aac";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(ADTSAudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A DV video stream:
{
// First, make sure that the RTPSinks' buffers will be large enough to handle the huge size of DV frames (as big as 288000).
OutPacketBuffer::maxSize = 2000000;
char const* streamName = "dvVideoTest";
char const* inputFileName = "test.dv";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(DVVideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A AC3 video elementary stream:
{
char const* streamName = "ac3AudioTest";
char const* inputFileName = "test.ac3";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(AC3AudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A Matroska ('.mkv') file, with video+audio+subtitle streams:
{
char const* streamName = "matroskaFileTest";
char const* inputFileName = "test.mkv";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
newDemuxWatchVariable = 0;
MatroskaFileServerDemux::createNew(*env, inputFileName, onMatroskaDemuxCreation, NULL);
env->taskScheduler().doEventLoop(&newDemuxWatchVariable);
Boolean sessionHasTracks = False;
ServerMediaSubsession* smss;
while ((smss = matroskaDemux->newServerMediaSubsession()) != NULL) {
sms->addSubsession(smss);
sessionHasTracks = True;
}
if (sessionHasTracks) {
rtspServer->addServerMediaSession(sms);
}
// otherwise, because the stream has no tracks, we don't add a ServerMediaSession to the server.
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A WebM ('.webm') file, with video(VP8)+audio(Vorbis) streams:
// (Note: ".webm' files are special types of Matroska files, so we use the same code as the Matroska ('.mkv') file code above.)
{
char const* streamName = "webmFileTest";
char const* inputFileName = "test.webm";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
newDemuxWatchVariable = 0;
MatroskaFileServerDemux::createNew(*env, inputFileName, onMatroskaDemuxCreation, NULL);
env->taskScheduler().doEventLoop(&newDemuxWatchVariable);
Boolean sessionHasTracks = False;
ServerMediaSubsession* smss;
while ((smss = matroskaDemux->newServerMediaSubsession()) != NULL) {
sms->addSubsession(smss);
sessionHasTracks = True;
}
if (sessionHasTracks) {
rtspServer->addServerMediaSession(sms);
}
// otherwise, because the stream has no tracks, we don't add a ServerMediaSession to the server.
announceStream(rtspServer, sms, streamName, inputFileName);
}
// An Ogg ('.ogg') file, with video and/or audio streams:
{
char const* streamName = "oggFileTest";
char const* inputFileName = "test.ogg";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
newDemuxWatchVariable = 0;
OggFileServerDemux::createNew(*env, inputFileName, onOggDemuxCreation, NULL);
env->taskScheduler().doEventLoop(&newDemuxWatchVariable);
Boolean sessionHasTracks = False;
ServerMediaSubsession* smss;
while ((smss = oggDemux->newServerMediaSubsession()) != NULL) {
sms->addSubsession(smss);
sessionHasTracks = True;
}
if (sessionHasTracks) {
rtspServer->addServerMediaSession(sms);
}
// otherwise, because the stream has no tracks, we don't add a ServerMediaSession to the server.
announceStream(rtspServer, sms, streamName, inputFileName);
}
// An Opus ('.opus') audio file:
// (Note: ".opus' files are special types of Ogg files, so we use the same code as the Ogg ('.ogg') file code above.)
{
char const* streamName = "opusFileTest";
char const* inputFileName = "test.opus";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
newDemuxWatchVariable = 0;
OggFileServerDemux::createNew(*env, inputFileName, onOggDemuxCreation, NULL);
env->taskScheduler().doEventLoop(&newDemuxWatchVariable);
Boolean sessionHasTracks = False;
ServerMediaSubsession* smss;
while ((smss = oggDemux->newServerMediaSubsession()) != NULL) {
sms->addSubsession(smss);
sessionHasTracks = True;
}
if (sessionHasTracks) {
rtspServer->addServerMediaSession(sms);
}
// otherwise, because the stream has no tracks, we don't add a ServerMediaSession to the server.
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A MPEG-2 Transport Stream, coming from a live UDP (raw-UDP or RTP/UDP) source:
{
char const* streamName = "mpeg2TransportStreamFromUDPSourceTest";
char const* inputAddressStr = "239.255.42.42";
// This causes the server to take its input from the stream sent by the "testMPEG2TransportStreamer" demo application.
// (Note: If the input UDP source is unicast rather than multicast, then change this to NULL.)
portNumBits const inputPortNum = 1234;
// This causes the server to take its input from the stream sent by the "testMPEG2TransportStreamer" demo application.
Boolean const inputStreamIsRawUDP = False;
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(MPEG2TransportUDPServerMediaSubsession
::createNew(*env, inputAddressStr, inputPortNum, inputStreamIsRawUDP));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "\n\"" << streamName << "\" stream, from a UDP Transport Stream input source \n\t(";
if (inputAddressStr != NULL) {
*env << "IP multicast address " << inputAddressStr << ",";
} else {
*env << "unicast;";
}
*env << " port " << inputPortNum << ")\n";
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
}
// Also, attempt to create a HTTP server for RTSP-over-HTTP tunneling.
// Try first with the default HTTP port (80), and then with the alternative HTTP
// port numbers (8000 and 8080).
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
return 0; // only to prevent compiler warning
}
int main(int argc, char** argv) {
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
UserAuthenticationDatabase* authDB = NULL;
#ifdef ACCESS_CONTROL
// To implement client access control to the RTSP server, do the following:
authDB = new UserAuthenticationDatabase;
authDB->addUserRecord("username1", "password1"); // replace these with real strings
// Repeat the above with each <username>, <password> that you wish to allow
// access to the server.
#endif
// Create the RTSP server:
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
char const* descriptionString
= "Session streamed by \"testOnDemandRTSPServer\"";
// Set up each of the possible streams that can be served by the
// RTSP server. Each such stream is implemented using a
// "ServerMediaSession" object, plus one or more
// "ServerMediaSubsession" objects for each audio/video substream.
// A MPEG-4 video elementary stream:
{
char const* streamName = "mpeg4ESVideoTest";
char const* inputFileName = "test.m4e";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(MPEG4VideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A H264 video elementary stream
{
/* char const* streamName = "h264VideoTest";
char const* inputFileName = "test.h264";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(H264VideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
*/
}
// A MPEG-1 or 2 audio+video program stream:
{
char const* streamName = "mpeg1or2AudioVideoTest";
char const* inputFileName = "test.mpg";
// NOTE: This *must* be a Program Stream; not an Elementary Stream
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
MPEG1or2FileServerDemux* demux
= MPEG1or2FileServerDemux::createNew(*env, inputFileName, reuseFirstSource);
sms->addSubsession(demux->newVideoServerMediaSubsession(iFramesOnly));
sms->addSubsession(demux->newAudioServerMediaSubsession());
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A MPEG-1 or 2 video elementary stream:
{
char const* streamName = "mpeg1or2ESVideoTest";
char const* inputFileName = "testv.mpg";
// NOTE: This *must* be a Video Elementary Stream; not a Program Stream
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(MPEG1or2VideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource, iFramesOnly));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A MP3 audio stream (actually, any MPEG-1 or 2 audio file will work):
// To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
//#define STREAM_USING_ADUS 1
// To also reorder ADUs before streaming, uncomment the following:
//#define INTERLEAVE_ADUS 1
// (For more information about ADUs and interleaving,
// see <http://www.live555.com/rtp-mp3/>)
{
char const* streamName = "mp3AudioTest";
char const* inputFileName = "test.mp3";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
Boolean useADUs = False;
Interleaving* interleaving = NULL;
#ifdef STREAM_USING_ADUS
useADUs = True;
#ifdef INTERLEAVE_ADUS
unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...
unsigned const interleaveCycleSize
= (sizeof interleaveCycle)/(sizeof (unsigned char));
interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
#endif
#endif
sms->addSubsession(MP3AudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource,
useADUs, interleaving));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A WAV audio stream:
{
char const* streamName = "wavAudioTest";
char const* inputFileName = "test.wav";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
// To convert 16-bit PCM data to 8-bit u-law, prior to streaming,
// change the following to True:
Boolean convertToULaw = False;
sms->addSubsession(WAVAudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource, convertToULaw));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// An AMR audio stream:
{
char const* streamName = "amrAudioTest";
char const* inputFileName = "test.amr";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(AMRAudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A 'VOB' file (e.g., from an unencrypted DVD):
{
char const* streamName = "vobTest";
char const* inputFileName = "test.vob";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
// Note: VOB files are MPEG-2 Program Stream files, but using AC-3 audio
MPEG1or2FileServerDemux* demux
= MPEG1or2FileServerDemux::createNew(*env, inputFileName, reuseFirstSource);
sms->addSubsession(demux->newVideoServerMediaSubsession(iFramesOnly));
sms->addSubsession(demux->newAC3AudioServerMediaSubsession());
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A MPEG-2 Transport Stream:
{
char const* streamName = "mpeg2TransportStreamTest";
char const* inputFileName = "test.ts";
char const* indexFileName = "test.tsx";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(MPEG2TransportFileServerMediaSubsession
::createNew(*env, inputFileName, indexFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// An AAC audio stream (ADTS-format file):
{
char const* streamName = "aacAudioTest";
char const* inputFileName = "test.aac";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(ADTSAudioFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
// A DV video stream:
{
// First, make sure that the RTPSinks' buffers will be large enough to handle the huge size of DV frames (as big as 288000).
OutPacketBuffer::maxSize = 300000;
char const* streamName = "dvVideoTest";
char const* inputFileName = "test.dv";
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
sms->addSubsession(DVVideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource));
rtspServer->addServerMediaSession(sms);
announceStream(rtspServer, sms, streamName, inputFileName);
}
env->taskScheduler().doEventLoop(); // does not return
return 0; // only to prevent compiler warning
}
int myRTSPServer(){
Boolean bFlag;
// Begin by setting up our usage environment:
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
UserAuthenticationDatabase* authDB = NULL;
// Create the RTSP server:
RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
}
char const* descriptionString
= "Session streamed by \"testOnDemandRTSPServer\"";
// Set up each of the possible streams that can be served by the
// RTSP server. Each such stream is implemented using a
// "ServerMediaSession" object, plus one or more
// "ServerMediaSubsession" objects for each audio/video substream.
// A H.264 video elementary stream:
{
char const* streamName = "BackChannelTest";
char const* inputFileName = "slamtv10.264";
char const* audioFileName = "slamtv10.aac";
char const* outputFileName = "receive.pcm";
reuseFirstSource = True;
// check if test file is exist
{
FILE *fp=NULL;
fp = fopen(inputFileName,"r");
if(fp==NULL) printf("File %s is not exist\n", inputFileName); else fclose(fp);
fp = fopen(audioFileName,"r");
if(fp==NULL) printf("File %s is not exist\n", audioFileName); else fclose(fp);
}
// Stream 1: H.264 video
ServerMediaSession* sms
= ServerMediaSession::createNew(*env, streamName, streamName,
descriptionString);
H264VideoFileServerMediaSubsession *sub =H264VideoFileServerMediaSubsession
::createNew(*env, inputFileName, reuseFirstSource);
bFlag = sms->addSubsession(sub);
if(bFlag==False) printf("addSubsession for %s error\n", inputFileName);
// Stream 2: AAC audio stream (ADTS-format file):
ADTSAudioFileServerMediaSubsession *sub2 =ADTSAudioFileServerMediaSubsession
::createNew(*env, audioFileName, reuseFirstSource);
bFlag = sms->addSubsession(sub2);
if(bFlag==False) printf("addSubsession for %s error\n", audioFileName);
// Stream 3: backchannel AAC audio
// TODO: modify here to support backchannel
// implement a new class named ADTSBackChannelAudioFileServerMediaSubsession
// use RTPSource to receive data and use ADTSAudioFileSink to save data to file
//ADTSBackChannelAudioFileServerMediaSubsession *sub3 =ADTSBackChannelAudioFileServerMediaSubsession
WaveBackChannelAudioFileServerMediaSubsession* sub3 = WaveBackChannelAudioFileServerMediaSubsession
::createNew(*env, outputFileName, reuseFirstSource);
sub3->setSubsessionAsBackChannel();
bFlag = sms->addSubsession(sub3);
if(bFlag==False) printf("addSubsession for %s error\n", outputFileName);
rtspServer->addServerMediaSession(sms);
// 20140703 albert.liao modified start
// we should notify OnDemandServerMediaSubsession or ServerMediaSubSession that we already create a backchannel subsession
// so that ServerMediaSubSession can do
// 1. create a SDP with backchannel
// 2. create a RTPSource to read data from RTPClient
// 3. create a FileSink to save received data to file
// 20140703 albert.liao modified end
announceStream(rtspServer, sms, streamName, inputFileName);
}
// Also, attempt to create a HTTP server for RTSP-over-HTTP tunneling.
// Try first with the default HTTP port (80), and then with the alternative HTTP
// port numbers (8000 and 8080).
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
return 0; // only to prevent compiler warning
}
