mirror of
https://github.com/mpromonet/v4l2rtspserver
synced 2024-11-11 19:10:40 +00:00
198 lines
8.0 KiB
C++
198 lines
8.0 KiB
C++
/* ---------------------------------------------------------------------------
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** This software is in the public domain, furnished "as is", without technical
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** support, and with no warranty, express or implied, as to its usefulness for
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** any purpose.
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**
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** HTTPServer.h
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**
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** V4L2 RTSP streamer
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**
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** HTTP server that serves HLS & MPEG-DASH playlist and segments
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**
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** -------------------------------------------------------------------------*/
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#include "RTSPServer.hh"
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#include "RTSPCommon.hh"
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#include <GroupsockHelper.hh> // for "ignoreSigPipeOnSocket()"
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#define TCP_STREAM_SINK_MIN_READ_SIZE 1000
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#define TCP_STREAM_SINK_BUFFER_SIZE 10000
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class TCPSink: public MediaSink {
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public:
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TCPSink(UsageEnvironment& env, int socketNum) : MediaSink(env), fUnwrittenBytesStart(0), fUnwrittenBytesEnd(0), fInputSourceIsOpen(False), fOutputSocketIsWritable(True),fOutputSocketNum(socketNum) {
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ignoreSigPipeOnSocket(socketNum);
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}
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protected:
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virtual ~TCPSink() {
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envir().taskScheduler().disableBackgroundHandling(fOutputSocketNum);
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}
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protected:
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virtual Boolean continuePlaying() {
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fInputSourceIsOpen = fSource != NULL;
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processBuffer();
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return True;
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}
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private:
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void processBuffer(){
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// First, try writing data to our output socket, if we can:
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if (fOutputSocketIsWritable && numUnwrittenBytes() > 0) {
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int numBytesWritten = send(fOutputSocketNum, (const char*)&fBuffer[fUnwrittenBytesStart], numUnwrittenBytes(), 0);
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if (numBytesWritten < (int)numUnwrittenBytes()) {
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// The output socket is no longer writable. Set a handler to be called when it becomes writable again.
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fOutputSocketIsWritable = False;
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if (envir().getErrno() != EPIPE) { // on this error, the socket might still be writable, but no longer usable
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envir().taskScheduler().setBackgroundHandling(fOutputSocketNum, SOCKET_WRITABLE, socketWritableHandler, this);
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}
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}
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if (numBytesWritten > 0) {
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// We wrote at least some of our data. Update our buffer pointers:
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fUnwrittenBytesStart += numBytesWritten;
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if (fUnwrittenBytesStart > fUnwrittenBytesEnd) fUnwrittenBytesStart = fUnwrittenBytesEnd; // sanity check
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if (fUnwrittenBytesStart == fUnwrittenBytesEnd && (!fInputSourceIsOpen || !fSource->isCurrentlyAwaitingData())) {
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fUnwrittenBytesStart = fUnwrittenBytesEnd = 0; // reset the buffer to empty
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}
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}
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}
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// Then, read from our input source, if we can (& we're not already reading from it):
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if (fInputSourceIsOpen && freeBufferSpace() >= TCP_STREAM_SINK_MIN_READ_SIZE && !fSource->isCurrentlyAwaitingData()) {
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fSource->getNextFrame(&fBuffer[fUnwrittenBytesEnd], freeBufferSpace(), afterGettingFrame, this, ourOnSourceClosure, this);
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} else if (!fInputSourceIsOpen && numUnwrittenBytes() == 0) {
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// We're now done:
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onSourceClosure();
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}
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}
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static void socketWritableHandler(void* clientData, int mask) { ((TCPSink*)clientData)->socketWritableHandler(); }
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void socketWritableHandler() {
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envir().taskScheduler().disableBackgroundHandling(fOutputSocketNum); // disable this handler until the next time it's needed
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fOutputSocketIsWritable = True;
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processBuffer();
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}
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static void afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,
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struct timeval presentationTime, unsigned durationInMicroseconds) {
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((TCPSink*)clientData)->afterGettingFrame(frameSize, numTruncatedBytes);
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}
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void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes) {
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if (numTruncatedBytes > 0) {
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envir() << "TCPStreamSink::afterGettingFrame(): The input frame data was too large for our buffer. "
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<< numTruncatedBytes
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<< " bytes of trailing data was dropped! Correct this by increasing the definition of \"TCP_STREAM_SINK_BUFFER_SIZE\" in \"include/TCPStreamSink.hh\".\n";
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}
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fUnwrittenBytesEnd += frameSize;
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processBuffer();
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}
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static void ourOnSourceClosure(void* clientData) { ((TCPSink*)clientData)->ourOnSourceClosure(); }
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void ourOnSourceClosure() {
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// The input source has closed:
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fInputSourceIsOpen = False;
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processBuffer();
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}
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unsigned numUnwrittenBytes() const { return fUnwrittenBytesEnd - fUnwrittenBytesStart; }
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unsigned freeBufferSpace() const { return TCP_STREAM_SINK_BUFFER_SIZE - fUnwrittenBytesEnd; }
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private:
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unsigned char fBuffer[TCP_STREAM_SINK_BUFFER_SIZE];
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unsigned fUnwrittenBytesStart, fUnwrittenBytesEnd;
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Boolean fInputSourceIsOpen, fOutputSocketIsWritable;
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int fOutputSocketNum;
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};
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// ---------------------------------------------------------
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// Extend RTSP server to add support for HLS and MPEG-DASH
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// ---------------------------------------------------------
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#if LIVEMEDIA_LIBRARY_VERSION_INT < 1606435200
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#define SOCKETCLIENT sockaddr_in
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#else
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#define SOCKETCLIENT sockaddr_storage
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#endif
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class HTTPServer : public RTSPServer
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{
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class HTTPClientConnection : public RTSPServer::RTSPClientConnection
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{
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public:
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HTTPClientConnection(RTSPServer& ourServer, int clientSocket, struct SOCKETCLIENT clientAddr)
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: 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:
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void sendHeader(const char* contentType, unsigned int contentLength);
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void streamSource(FramedSource* source);
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void streamSource(const std::string & content);
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ServerMediaSubsession* getSubsesion(const char* urlSuffix);
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bool sendFile(char const* urlSuffix);
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bool sendM3u8PlayList(char const* urlSuffix);
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bool sendMpdPlayList(char const* urlSuffix);
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virtual void handleHTTPCmd_StreamingGET(char const* urlSuffix, char const* fullRequestStr);
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virtual void handleCmd_notFound();
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static void afterStreaming(void* clientData);
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private:
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static u_int32_t m_ClientSessionId;
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TCPSink* m_TCPSink;
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void* m_StreamToken;
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ServerMediaSubsession* m_Subsession;
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FramedSource* m_Source;
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};
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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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{
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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
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int ourSocketIPv6 = -1;
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#else
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int ourSocketIPv6 = setUpOurSocket(env, rtspPort, AF_INET6);
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#endif
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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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}
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return httpServer;
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}
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#if LIVEMEDIA_LIBRARY_VERSION_INT < 1611187200
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HTTPServer(UsageEnvironment& env, int ourSocketIPv4, int ourSocketIPv6, Port rtspPort, UserAuthenticationDatabase* authDatabase, unsigned reclamationTestSeconds, unsigned int hlsSegment, const std::string & webroot)
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: RTSPServer(env, ourSocketIPv4, rtspPort, authDatabase, reclamationTestSeconds), m_hlsSegment(hlsSegment), m_webroot(webroot)
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#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)
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: RTSPServer(env, ourSocketIPv4, ourSocketIPv6, rtspPort, authDatabase, reclamationTestSeconds), m_hlsSegment(hlsSegment), m_webroot(webroot)
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#endif
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{
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if ( (!m_webroot.empty()) && (*m_webroot.rend() != '/') ) {
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m_webroot += "/";
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}
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}
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RTSPServer::RTSPClientConnection* createNewClientConnection(int clientSocket, struct SOCKETCLIENT clientAddr)
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{
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return new HTTPClientConnection(*this, clientSocket, clientAddr);
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}
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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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};
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