v4l2rtspserver/inc/HTTPServer.h

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/* ---------------------------------------------------------------------------
** This software is in the public domain, furnished "as is", without technical
** support, and with no warranty, express or implied, as to its usefulness for
** any purpose.
**
** HTTPServer.h
**
** V4L2 RTSP streamer
**
** HTTP server that serves HLS & MPEG-DASH playlist and segments
**
** -------------------------------------------------------------------------*/
#include "RTSPServer.hh"
#include "RTSPCommon.hh"
#include <GroupsockHelper.hh> // for "ignoreSigPipeOnSocket()"
#define TCP_STREAM_SINK_MIN_READ_SIZE 1000
#define TCP_STREAM_SINK_BUFFER_SIZE 10000
class TCPSink: public MediaSink {
public:
TCPSink(UsageEnvironment& env, int socketNum) : MediaSink(env), fUnwrittenBytesStart(0), fUnwrittenBytesEnd(0), fInputSourceIsOpen(False), fOutputSocketIsWritable(True),fOutputSocketNum(socketNum) {
ignoreSigPipeOnSocket(socketNum);
}
protected:
virtual ~TCPSink() {
envir().taskScheduler().disableBackgroundHandling(fOutputSocketNum);
}
protected:
virtual Boolean continuePlaying() {
fInputSourceIsOpen = fSource != NULL;
processBuffer();
return True;
}
private:
void processBuffer(){
// First, try writing data to our output socket, if we can:
if (fOutputSocketIsWritable && numUnwrittenBytes() > 0) {
int numBytesWritten = send(fOutputSocketNum, (const char*)&fBuffer[fUnwrittenBytesStart], numUnwrittenBytes(), 0);
if (numBytesWritten < (int)numUnwrittenBytes()) {
// The output socket is no longer writable. Set a handler to be called when it becomes writable again.
fOutputSocketIsWritable = False;
if (envir().getErrno() != EPIPE) { // on this error, the socket might still be writable, but no longer usable
envir().taskScheduler().setBackgroundHandling(fOutputSocketNum, SOCKET_WRITABLE, socketWritableHandler, this);
}
}
if (numBytesWritten > 0) {
// We wrote at least some of our data. Update our buffer pointers:
fUnwrittenBytesStart += numBytesWritten;
if (fUnwrittenBytesStart > fUnwrittenBytesEnd) fUnwrittenBytesStart = fUnwrittenBytesEnd; // sanity check
if (fUnwrittenBytesStart == fUnwrittenBytesEnd && (!fInputSourceIsOpen || !fSource->isCurrentlyAwaitingData())) {
fUnwrittenBytesStart = fUnwrittenBytesEnd = 0; // reset the buffer to empty
}
}
}
// Then, read from our input source, if we can (& we're not already reading from it):
if (fInputSourceIsOpen && freeBufferSpace() >= TCP_STREAM_SINK_MIN_READ_SIZE && !fSource->isCurrentlyAwaitingData()) {
fSource->getNextFrame(&fBuffer[fUnwrittenBytesEnd], freeBufferSpace(), afterGettingFrame, this, ourOnSourceClosure, this);
} else if (!fInputSourceIsOpen && numUnwrittenBytes() == 0) {
// We're now done:
onSourceClosure();
}
}
static void socketWritableHandler(void* clientData, int mask) { ((TCPSink*)clientData)->socketWritableHandler(); }
void socketWritableHandler() {
envir().taskScheduler().disableBackgroundHandling(fOutputSocketNum); // disable this handler until the next time it's needed
fOutputSocketIsWritable = True;
processBuffer();
}
static void afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,
struct timeval presentationTime, unsigned durationInMicroseconds) {
((TCPSink*)clientData)->afterGettingFrame(frameSize, numTruncatedBytes);
}
void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes) {
if (numTruncatedBytes > 0) {
envir() << "TCPStreamSink::afterGettingFrame(): The input frame data was too large for our buffer. "
<< numTruncatedBytes
<< " bytes of trailing data was dropped! Correct this by increasing the definition of \"TCP_STREAM_SINK_BUFFER_SIZE\" in \"include/TCPStreamSink.hh\".\n";
}
fUnwrittenBytesEnd += frameSize;
processBuffer();
}
static void ourOnSourceClosure(void* clientData) { ((TCPSink*)clientData)->ourOnSourceClosure(); }
void ourOnSourceClosure() {
// The input source has closed:
fInputSourceIsOpen = False;
processBuffer();
}
unsigned numUnwrittenBytes() const { return fUnwrittenBytesEnd - fUnwrittenBytesStart; }
unsigned freeBufferSpace() const { return TCP_STREAM_SINK_BUFFER_SIZE - fUnwrittenBytesEnd; }
private:
unsigned char fBuffer[TCP_STREAM_SINK_BUFFER_SIZE];
unsigned fUnwrittenBytesStart, fUnwrittenBytesEnd;
Boolean fInputSourceIsOpen, fOutputSocketIsWritable;
int fOutputSocketNum;
};
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// ---------------------------------------------------------
// Extend RTSP server to add support for HLS and MPEG-DASH
// ---------------------------------------------------------
#if LIVEMEDIA_LIBRARY_VERSION_INT < 1606435200
#define SOCKETCLIENT sockaddr_in
#else
#define SOCKETCLIENT sockaddr_storage const&
#endif
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class HTTPServer : public RTSPServer
{
class HTTPClientConnection : public RTSPServer::RTSPClientConnection
{
public:
HTTPClientConnection(RTSPServer& ourServer, int clientSocket, struct SOCKETCLIENT clientAddr)
: RTSPServer::RTSPClientConnection(ourServer, clientSocket, clientAddr), m_TCPSink(NULL), m_StreamToken(NULL), m_Subsession(NULL), m_Source(NULL) {
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}
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virtual ~HTTPClientConnection();
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private:
void sendHeader(const char* contentType, unsigned int contentLength);
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void streamSource(FramedSource* source);
void streamSource(const std::string & content);
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ServerMediaSubsession* getSubsesion(const char* urlSuffix);
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bool sendFile(char const* urlSuffix);
bool sendM3u8PlayList(char const* urlSuffix);
bool sendMpdPlayList(char const* urlSuffix);
virtual void handleHTTPCmd_StreamingGET(char const* urlSuffix, char const* fullRequestStr);
virtual void handleCmd_notFound();
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static void afterStreaming(void* clientData);
private:
static u_int32_t m_ClientSessionId;
TCPSink* m_TCPSink;
void* m_StreamToken;
ServerMediaSubsession* m_Subsession;
FramedSource* m_Source;
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};
public:
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static HTTPServer* createNew(UsageEnvironment& env, Port rtspPort, UserAuthenticationDatabase* authDatabase, unsigned reclamationTestSeconds, unsigned int hlsSegment, const std::string webroot)
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{
HTTPServer* httpServer = NULL;
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#if LIVEMEDIA_LIBRARY_VERSION_INT < 1610928000
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int ourSocketIPv4 = setUpOurSocket(env, rtspPort);
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#else
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int ourSocketIPv4 = setUpOurSocket(env, rtspPort, AF_INET);
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#endif
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#if LIVEMEDIA_LIBRARY_VERSION_INT < 1611187200
int ourSocketIPv6 = -1;
#else
int ourSocketIPv6 = setUpOurSocket(env, rtspPort, AF_INET6);
#endif
if (ourSocketIPv4 != -1)
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{
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httpServer = new HTTPServer(env, ourSocketIPv4, ourSocketIPv6, rtspPort, authDatabase, reclamationTestSeconds, hlsSegment, webroot);
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}
return httpServer;
}
#if LIVEMEDIA_LIBRARY_VERSION_INT < 1611187200
HTTPServer(UsageEnvironment& env, int ourSocketIPv4, int ourSocketIPv6, Port rtspPort, UserAuthenticationDatabase* authDatabase, unsigned reclamationTestSeconds, unsigned int hlsSegment, const std::string & webroot)
: RTSPServer(env, ourSocketIPv4, rtspPort, authDatabase, reclamationTestSeconds), m_hlsSegment(hlsSegment), m_webroot(webroot)
#else
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HTTPServer(UsageEnvironment& env, int ourSocketIPv4, int ourSocketIPv6, Port rtspPort, UserAuthenticationDatabase* authDatabase, unsigned reclamationTestSeconds, unsigned int hlsSegment, const std::string & webroot)
: RTSPServer(env, ourSocketIPv4, ourSocketIPv6, rtspPort, authDatabase, reclamationTestSeconds), m_hlsSegment(hlsSegment), m_webroot(webroot)
#endif
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{
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if ( (!m_webroot.empty()) && (*m_webroot.rend() != '/') ) {
m_webroot += "/";
}
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}
RTSPServer::RTSPClientConnection* createNewClientConnection(int clientSocket, struct SOCKETCLIENT clientAddr)
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{
return new HTTPClientConnection(*this, clientSocket, clientAddr);
}
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private:
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const unsigned int m_hlsSegment;
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std::string m_webroot;
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};