// TestDemandRtspServer.cpp.cpp : 定义控制台应用程序的入口点。
//
#include "stdafx.h"
//int _tmain(int argc, _TCHAR* argv[])
//{
// return 0;
//}
/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 2.1 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for
more details.
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// Copyright (c) 1996-2013, Live Networks, Inc. All rights reserved
// A test program that demonstrates how to stream - via unicast RTP
// - various kinds of file on demand, using a built-in RTSP server.
// main program
#include "liveMedia.hh"
#include "BasicUsageEnvironment.hh"
UsageEnvironment* env;
// To make the second and subsequent client for each stream reuse the same
// input stream as the first client (rather than playing the file from the
// start for each client), change the following "False" to "True":
Boolean reuseFirstSource = False;
// To stream *only* MPEG-1 or 2 video "I" frames
// (e.g., to reduce network bandwidth),
// change the following "False" to "True":
Boolean iFramesOnly = False;
static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
char const* streamName, char const* inputFileName); // fwd
static char newMatroskaDemuxWatchVariable;
static MatroskaFileServerDemux* demux;
static void onMatroskaDemuxCreation(MatroskaFileServerDemux* newDemux, void* /*clientData*/) {
demux = newDemux;
newMatroskaDemuxWatchVariable = 1;
}
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 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);
}
// 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);
newMatroskaDemuxWatchVariable = 0;
MatroskaFileServerDemux::createNew(*env, inputFileName, onMatroskaDemuxCreation, NULL);
env->taskScheduler().doEventLoop(&newMatroskaDemuxWatchVariable);
Boolean sessionHasTracks = False;
ServerMediaSubsession* smss;
while ((smss = demux->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);
newMatroskaDemuxWatchVariable = 0;
MatroskaFileServerDemux::createNew(*env, inputFileName, onMatroskaDemuxCreation, NULL);
env->taskScheduler().doEventLoop(&newMatroskaDemuxWatchVariable);
Boolean sessionHasTracks = False;
ServerMediaSubsession* smss;
while ((smss = demux->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
}
static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
char const* streamName, char const* inputFileName) {
char* url = rtspServer->rtspURL(sms);
UsageEnvironment& env = rtspServer->envir();
env << "\n\"" << streamName << "\" stream, from the file \""
<< inputFileName << "\"\n";
env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
}