我想从文件中进行 RTSP 流式传输,所以我使用了 live 555 库。在 live 555 测试目录中“testMpeg2TransportStreamer”程序流式传输 TS 格式文件,我所做的是,我将整个代码放在线程中,所以每当客户端请求流式传输线程开始工作,当客户端说 DONT STREAM 然后线程在关闭线程时关闭我还编写了代码 Medium::close(rtsp 服务器的指针),所以要关闭 rtsp 服务器,这适用于第一个 STREAM 和 DONT STREAM 请求,但在 DONT STREAM 请求之后,当客户端对 STREAM 说时,我调试了代码,发现它无法创建 rtsp 服务器。然后我使用了另一种方法,在第一个 STREAM 请求时只创建一次 rtsp 服务器,而不是在我的整个程序退出之前关闭它,但这也失败了。
#define TRANSPORT_PACKET_SIZE 188
#define TRANSPORT_PACKETS_PER_NETWORK_PACKET 7
#define IMPLEMENT_RTSP_SERVER
// To stream using "source-specific multicast" (SSM), uncomment the following:
//#define USE_SSM 1
#ifdef USE_SSM
Boolean const isSSM = True;
#else
Boolean const isSSM = False;
#endif
/********************Global variable***************************/
UsageEnvironment* env=NULL;
FramedSource* videoSource;
RTPSink* videoSink;
DeviceSourceFICard* fileSource;
FICardDeviceParameters fi_params;
HANDLE g_hRtpComThread;
DWORD g_dwRtpComThreadID;
char g_ExitEventLoop;
void play(); // forward
RTSPServer* rtspServer=NULL;
ServerMediaSession* sms;
int initLm555Settings(void)
{
scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
destinationAddressStr//make it global
#ifdef USE_SSM
= "232.255.42.42";
#else
= "239.255.42.42";
const unsigned short rtpPortNum = 18888;//make it global
rtpPortNum1=rtpPortNum;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 7;
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
rtpGroupsock.multicastSendOnly();
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);
rtcpGroupsock.multicastSendOnly();
#ifdef USE_SSM
rtpGroupsock.multicastSendOnly();
rtcpGroupsock.multicastSendOnly();
#endif
g_ExitEventLoop = 0;
videoSink =
SimpleRTPSink::createNew(*env, &rtpGroupsock, 33, 90000, "video", "MP2T",
1, True, False /*no 'M' bit*/);
setSendBufferTo(*env, rtpGroupsock.socketNum(), 1024 * 1024);
setSendBufferTo(*env, rtpGroupsock.socketNum(), 1024 * 1024);
// 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 */, isSSM);
UserAuthenticationDatabase* authDB = NULL;
portNumBits rtspServerPortNum = 554;
unsigned reclamationTestSeconds=65U;
rtspServer = RTSPServer::createNew(*env,rtspServerPortNum, authDB, reclamationTestSeconds);
if (rtspServer == NULL)
{
*env << "Failed to create RTSP server: " <<env->getResultMsg()<<"\n";
rtspServerPortNum = 8554;
rtspServer = RTSPServer::createNew(*env,rtspServerPortNum);
if (rtspServer == NULL)
{
return 0;
}
Boolean const inputStreamIsRawUDP = False;
char const* descriptionString={"Session streamed by \"testOnDemandRT\""};
sms=ServerMediaSession::createNew(*env, streamName, streamName,descriptionString);
sms->addSubsession(MPEG2TransportUDPServerMediaSubsession::createNew(*env,destinationAddressStr,rtpPortNum1,inputStreamIsRawUDP));
rtspServer->addServerMediaSession(sms);
char* url = rtspServer->rtspURL(sms);
*env << "Play this stream using the URL \"" << url << "\"\n";
delete[] url;
if (rtspServer->setUpTunnelingOverHTTP(sport) || rtspServer->setUpTunnelingOverHTTP(sport) || rtspServer->setUpTunnelingOverHTTP(sport))
{
out<<"\n\n\n(We use port "<<rtspServer->httpServerPortNum()<<" for optional RTSP-over-HTTP tunneling.)\n";
}
else
{
pDailyLogger->LogInfoString("(RTSP-over-HTTP tunneling is not available.)");
cout<<"\n\n\n(RTSP-over-HTTP tunneling is not available.)";
}
play();
env->taskScheduler().doEventLoop(&g_ExitEventLoop);
Medium::close(rtspServer);
Medium::close(rtcp);
Medium::close(videoSink);
rtpGroupsock.removeAllDestinations();
rtcpGroupsock.removeAllDestinations();
env->reclaim();
delete scheduler;
pDailyLogger->LogDebugString("OUT::initLm555Settings Thread");
return 0; // only to prevent compiler warning
}
void afterPlaying(void* /*clientData*/) {
*env << "...done reading from file\n";
videoSink->stopPlaying();
Medium::close(videoSource);
play();
}
void play() {
// Open the input file as a 'byte-stream file source':
fi_params.nFICardFrameSize = TRANSPORT_PACKETS_PER_NETWORK_PACKET * TRANSPORT_PACKET_SIZE;
fi_params.pfnGetRTPPayload = GetRTPPayload;
fi_params.socketNum = videoSink->groupsockBeingUsed().socketNum();
DeviceParameters temp;
fileSource = DeviceSourceFICard::createNew(*env, fi_params, temp);
if (fileSource == NULL) {
*env << "Unable to open Foresight card as a byte-stream file source\n";
exit(1);
}
FramedSource* videoES = fileSource;
videoSource = MPEG1or2VideoStreamDiscreteFramer::createNew(*env, videoES);
*env << "Beginning to read from file...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
}
void StartRTPProcess(void)
{
g_hRtpComThread = CreateThread((LPSECURITY_ATTRIBUTES) NULL, 0,
(LPTHREAD_START_ROUTINE)initLm555Settings, 0, 0, &g_dwRtpComThreadID);
if(g_hRtpComThread) SetThreadPriority(g_hRtpComThread, THREAD_PRIORITY_LOWEST/*THREAD_PRIORITY_NORMAL*/);
}
int StopRTProcess(void)
{
g_ExitEventLoop = 1;
g_ExitEventLoop = 0;
g_hRtpComThread = 0;
g_dwRtpComThreadID = 0;
}
当我收到 DONT STREAM 消息时,Sir StopRTProcess() 被调用,我错过了什么请告诉