/**********
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
**********/
// "liveMedia"
// Copyright (c) 1996-2011 Live Networks, Inc.  All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand.
// Implementation

#include "OnDemandServerMediaSubsession.hh"
#include "RTCP.hh"
#include "BasicUDPSink.hh"
#include <GroupsockHelper.hh>

OnDemandServerMediaSubsession
::OnDemandServerMediaSubsession(UsageEnvironment& env,
                Boolean reuseFirstSource,
                portNumBits initialPortNum)
  : ServerMediaSubsession(env),
    fSDPLines(NULL), fReuseFirstSource(reuseFirstSource), fInitialPortNum(initialPortNum), fLastStreamToken(NULL) {
  fDestinationsHashTable = HashTableLive555::create(ONE_WORD_HASH_KEYS);
  gethostname(fCNAME, sizeof fCNAME);
  fCNAME[sizeof fCNAME-1] = '\0'; // just in case
}

class Destinations {
public:
  Destinations(struct in_addr const& destAddr,
           Port const& rtpDestPort,
           Port const& rtcpDestPort)
    : isTCP(False), addr(destAddr), rtpPort(rtpDestPort), rtcpPort(rtcpDestPort) {
  }
  Destinations(int tcpSockNum, unsigned char rtpChanId, unsigned char rtcpChanId)
    : isTCP(True), rtpPort(0) /*dummy*/, rtcpPort(0) /*dummy*/,
      tcpSocketNum(tcpSockNum), rtpChannelId(rtpChanId), rtcpChannelId(rtcpChanId) {
  }

public:
  Boolean isTCP;
  struct in_addr addr;
  Port rtpPort;
  Port rtcpPort;
  int tcpSocketNum;
  unsigned char rtpChannelId, rtcpChannelId;
};

OnDemandServerMediaSubsession::~OnDemandServerMediaSubsession() {
  delete[] fSDPLines;

  // Clean out the destinations hash table:
  while (1) {
    Destinations* destinations
      = (Destinations*)(fDestinationsHashTable->RemoveNext());
    if (destinations == NULL) break;
    delete destinations;
  }
  delete fDestinationsHashTable;
}

char const*
OnDemandServerMediaSubsession::sdpLines() {
  if (fSDPLines == NULL) {
    // We need to construct a set of SDP lines that describe this
    // subsession (as a unicast stream).  To do so, we first create
    // dummy (unused) source and "RTPSink" objects,
    // whose parameters we use for the SDP lines:
    unsigned estBitrate;
    FramedSource* inputSource = createNewStreamSource(0, estBitrate);
    if (inputSource == NULL) return NULL; // file not found

    struct in_addr dummyAddr;
    dummyAddr.s_addr = 0;
    Groupsock dummyGroupsock(envir(), dummyAddr, 0, 0);
    unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
    RTPSink* dummyRTPSink
      = createNewRTPSink(&dummyGroupsock, rtpPayloadType, inputSource);

    setSDPLinesFromRTPSink(dummyRTPSink, inputSource, estBitrate);
    Medium::close(dummyRTPSink);
    closeStreamSource(inputSource);
  }

  return fSDPLines;
}

// A class that represents the state of an ongoing stream
class StreamState {
public:
  StreamState(OnDemandServerMediaSubsession& master,
              Port const& serverRTPPort, Port const& serverRTCPPort,
          RTPSink* rtpSink, BasicUDPSink* udpSink,
          unsigned totalBW, FramedSource* mediaSource,
          Groupsock* rtpGS, Groupsock* rtcpGS);
  virtual ~StreamState();

  void startPlaying(Destinations* destinations,
            TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
            ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
                    void* serverRequestAlternativeByteHandlerClientData);
  void pause();
  void endPlaying(Destinations* destinations);
  void reclaim();

  unsigned& referenceCount() { return fReferenceCount; }

  Port const& serverRTPPort() const { return fServerRTPPort; }
  Port const& serverRTCPPort() const { return fServerRTCPPort; }

  RTPSink* rtpSink() const { return fRTPSink; }

  float streamDuration() const { return fStreamDuration; }

  FramedSource* mediaSource() const { return fMediaSource; }

private:
  OnDemandServerMediaSubsession& fMaster;
  Boolean fAreCurrentlyPlaying;
  unsigned fReferenceCount;

  Port fServerRTPPort, fServerRTCPPort;

  RTPSink* fRTPSink;
  BasicUDPSink* fUDPSink;

  float fStreamDuration;
  unsigned fTotalBW; RTCPInstance* fRTCPInstance;

  FramedSource* fMediaSource;

  Groupsock* fRTPgs; Groupsock* fRTCPgs;
};

void OnDemandServerMediaSubsession
::getStreamParameters(unsigned clientSessionId,
              netAddressBits clientAddress,
              Port const& clientRTPPort,
              Port const& clientRTCPPort,
              int tcpSocketNum,
              unsigned char rtpChannelId,
              unsigned char rtcpChannelId,
              netAddressBits& destinationAddress,
              u_int8_t& /*destinationTTL*/,
              Boolean& isMulticast,
              Port& serverRTPPort,
              Port& serverRTCPPort,
              void*& streamToken) {
  if (destinationAddress == 0) destinationAddress = clientAddress;
  struct in_addr destinationAddr; destinationAddr.s_addr = destinationAddress;
  isMulticast = False;

  if (fLastStreamToken != NULL && fReuseFirstSource) {
    // Special case: Rather than creating a new 'StreamState',
    // we reuse the one that we've already created:
    serverRTPPort = ((StreamState*)fLastStreamToken)->serverRTPPort();
    serverRTCPPort = ((StreamState*)fLastStreamToken)->serverRTCPPort();
    ++((StreamState*)fLastStreamToken)->referenceCount();
    streamToken = fLastStreamToken;
  } else {
    // Normal case: Create a new media source:
    unsigned streamBitrate;
    FramedSource* mediaSource
      = createNewStreamSource(clientSessionId, streamBitrate);

    // Create 'groupsock' and 'sink' objects for the destination,
    // using previously unused server port numbers:
    RTPSink* rtpSink;
    BasicUDPSink* udpSink;
    Groupsock* rtpGroupsock;
    Groupsock* rtcpGroupsock;
    portNumBits serverPortNum;
    if (clientRTCPPort.num() == 0) {
      // We're streaming raw UDP (not RTP). Create a single groupsock:
      NoReuse dummy; // ensures that we skip over ports that are already in use
      for (serverPortNum = fInitialPortNum; ; ++serverPortNum) {
    struct in_addr dummyAddr; dummyAddr.s_addr = 0;

    serverRTPPort = serverPortNum;
    rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
    if (rtpGroupsock->socketNum() >= 0) break; // success
      }

      rtcpGroupsock = NULL;
      rtpSink = NULL;
      udpSink = BasicUDPSink::createNew(envir(), rtpGroupsock);
    } else {
      // Normal case: We're streaming RTP (over UDP or TCP).  Create a pair of
      // groupsocks (RTP and RTCP), with adjacent port numbers (RTP port number even):
      NoReuse dummy; // ensures that we skip over ports that are already in use
      for (portNumBits serverPortNum = fInitialPortNum; ; serverPortNum += 2) {
    struct in_addr dummyAddr; dummyAddr.s_addr = 0;

    serverRTPPort = serverPortNum;
    rtpGroupsock = new Groupsock(envir(), dummyAddr, serverRTPPort, 255);
    if (rtpGroupsock->socketNum() < 0) {
      delete rtpGroupsock;
      continue; // try again
    }

    serverRTCPPort = serverPortNum+1;
    rtcpGroupsock = new Groupsock(envir(), dummyAddr, serverRTCPPort, 255);
    if (rtcpGroupsock->socketNum() < 0) {
      delete rtpGroupsock;
      delete rtcpGroupsock;
      continue; // try again
    }

    break; // success
      }

      unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
      rtpSink = createNewRTPSink(rtpGroupsock, rtpPayloadType, mediaSource);
      udpSink = NULL;
    }

    // Turn off the destinations for each groupsock.  They'll get set later
    // (unless TCP is used instead):
    if (rtpGroupsock != NULL) rtpGroupsock->removeAllDestinations();
    if (rtcpGroupsock != NULL) rtcpGroupsock->removeAllDestinations();

    if (rtpGroupsock != NULL) {
      // Try to use a big send buffer for RTP -  at least 0.1 second of
      // specified bandwidth and at least 50 KB
      unsigned rtpBufSize = streamBitrate * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
      if (rtpBufSize < 50 * 1024) rtpBufSize = 50 * 1024;
      increaseSendBufferTo(envir(), rtpGroupsock->socketNum(), rtpBufSize);
    }

    // Set up the state of the stream.  The stream will get started later:
    streamToken = fLastStreamToken
      = new StreamState(*this, serverRTPPort, serverRTCPPort, rtpSink, udpSink,
            streamBitrate, mediaSource,
            rtpGroupsock, rtcpGroupsock);
  }

  // Record these destinations as being for this client session id:
  Destinations* destinations;
  if (tcpSocketNum < 0) { // UDP
    destinations = new Destinations(destinationAddr, clientRTPPort, clientRTCPPort);
  } else { // TCP
    destinations = new Destinations(tcpSocketNum, rtpChannelId, rtcpChannelId);
  }
  fDestinationsHashTable->Add((char const*)clientSessionId, destinations);
}

void OnDemandServerMediaSubsession::startStream(unsigned clientSessionId,
                        void* streamToken,
                        TaskFunc* rtcpRRHandler,
                        void* rtcpRRHandlerClientData,
                        unsigned short& rtpSeqNum,
                        unsigned& rtpTimestamp,
                        ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
                        void* serverRequestAlternativeByteHandlerClientData) {
  StreamState* streamState = (StreamState*)streamToken;
  Destinations* destinations
    = (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));
  if (streamState != NULL) {
    streamState->startPlaying(destinations,
                  rtcpRRHandler, rtcpRRHandlerClientData,
                  serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
    if (streamState->rtpSink() != NULL) {
      rtpSeqNum = streamState->rtpSink()->currentSeqNo();
      rtpTimestamp = streamState->rtpSink()->presetNextTimestamp();
    }
  }
}

void OnDemandServerMediaSubsession::pauseStream(unsigned /*clientSessionId*/,
                        void* streamToken) {
  // Pausing isn't allowed if multiple clients are receiving data from
  // the same source:
  if (fReuseFirstSource) return;

  StreamState* streamState = (StreamState*)streamToken;
  if (streamState != NULL) streamState->pause();
}

void OnDemandServerMediaSubsession::seekStream(unsigned /*clientSessionId*/,
                           void* streamToken, double seekNPT, double streamDuration) {
  // Seeking isn't allowed if multiple clients are receiving data from
  // the same source:
  if (fReuseFirstSource) return;

  StreamState* streamState = (StreamState*)streamToken;
  if (streamState != NULL && streamState->mediaSource() != NULL) {
    seekStreamSource(streamState->mediaSource(), seekNPT, streamDuration);
  }
}

void OnDemandServerMediaSubsession::setStreamScale(unsigned /*clientSessionId*/,
                           void* streamToken, float scale) {
  // Changing the scale factor isn't allowed if multiple clients are receiving data
  // from the same source:
  if (fReuseFirstSource) return;

  StreamState* streamState = (StreamState*)streamToken;
  if (streamState != NULL && streamState->mediaSource() != NULL) {
    setStreamSourceScale(streamState->mediaSource(), scale);
  }
}

void OnDemandServerMediaSubsession::deleteStream(unsigned clientSessionId,
                         void*& streamToken) {
  StreamState* streamState = (StreamState*)streamToken;

  // Look up (and remove) the destinations for this client session:
  Destinations* destinations
    = (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));
  if (destinations != NULL) {
    fDestinationsHashTable->Remove((char const*)clientSessionId);

    // Stop streaming to these destinations:
    if (streamState != NULL) streamState->endPlaying(destinations);
  }

  // Delete the "StreamState" structure if it's no longer being used:
  if (streamState != NULL) {
    if (streamState->referenceCount() > 0) --streamState->referenceCount();
    if (streamState->referenceCount() == 0) {
      delete streamState;
      streamToken = NULL;
    }
  }

  // Finally, delete the destinations themselves:
  delete destinations;
}

char const* OnDemandServerMediaSubsession
::getAuxSDPLine(RTPSink* rtpSink, FramedSource* /*inputSource*/) {
  // Default implementation:
  return rtpSink == NULL ? NULL : rtpSink->auxSDPLine();
}

void OnDemandServerMediaSubsession::seekStreamSource(FramedSource* /*inputSource*/,
                             double /*seekNPT*/, double /*streamDuration*/) {
  // Default implementation: Do nothing
}

void OnDemandServerMediaSubsession
::setStreamSourceScale(FramedSource* /*inputSource*/, float /*scale*/) {
  // Default implementation: Do nothing
}

void OnDemandServerMediaSubsession::closeStreamSource(FramedSource *inputSource) {
  Medium::close(inputSource);
}

void OnDemandServerMediaSubsession
::setSDPLinesFromRTPSink(RTPSink* rtpSink, FramedSource* inputSource, unsigned estBitrate) {
  if (rtpSink == NULL) return;

  char const* mediaType = rtpSink->sdpMediaType();
  unsigned char rtpPayloadType = rtpSink->rtpPayloadType();
  struct in_addr serverAddrForSDP; serverAddrForSDP.s_addr = fServerAddressForSDP;
  char* const ipAddressStr = strDup(our_inet_ntoa(serverAddrForSDP));
  char* rtpmapLine = rtpSink->rtpmapLine();
  char const* rangeLine = rangeSDPLine();
  char const* auxSDPLine = getAuxSDPLine(rtpSink, inputSource);
  if (auxSDPLine == NULL) auxSDPLine = "";

  char const* const sdpFmt =
    "m=%s %u RTP/AVP %d\r\n"
    "c=IN IP4 %s\r\n"
    "b=AS:%u\r\n"
    "%s"
    "%s"
    "%s"
    "a=control:%s\r\n";
  unsigned sdpFmtSize = strlen(sdpFmt)
    + strlen(mediaType) + 5 /* max short len */ + 3 /* max char len */
    + strlen(ipAddressStr)
    + 20 /* max int len */
    + strlen(rtpmapLine)
    + strlen(rangeLine)
    + strlen(auxSDPLine)
    + strlen(trackId());
  char* sdpLines = new char[sdpFmtSize];
  sprintf(sdpLines, sdpFmt,
      mediaType, // m= <media>
      fPortNumForSDP, // m= <port>
      rtpPayloadType, // m= <fmt list>
      ipAddressStr, // c= address
      estBitrate, // b=AS:<bandwidth>
      rtpmapLine, // a=rtpmap:... (if present)
      rangeLine, // a=range:... (if present)
      auxSDPLine, // optional extra SDP line
      trackId()); // a=control:<track-id>
  delete[] (char*)rangeLine; delete[] rtpmapLine; delete[] ipAddressStr;

  fSDPLines = strDup(sdpLines);
  delete[] sdpLines;
}


////////// StreamState implementation //////////

static void afterPlayingStreamState(void* clientData) {
  StreamState* streamState = (StreamState*)clientData;
  if (streamState->streamDuration() == 0.0) {
    // When the input stream ends, tear it down.  This will cause a RTCP "BYE"
    // to be sent to each client, teling it that the stream has ended.
    // (Because the stream didn't have a known duration, there was no other
    //  way for clients to know when the stream ended.)
    streamState->reclaim();
  }
  // Otherwise, keep the stream alive, in case a client wants to
  // subsequently re-play the stream starting from somewhere other than the end.
  // (This can be done only on streams that have a known duration.)
}

StreamState::StreamState(OnDemandServerMediaSubsession& master,
                         Port const& serverRTPPort, Port const& serverRTCPPort,
             RTPSink* rtpSink, BasicUDPSink* udpSink,
             unsigned totalBW, FramedSource* mediaSource,
             Groupsock* rtpGS, Groupsock* rtcpGS)
  : fMaster(master), fAreCurrentlyPlaying(False), fReferenceCount(1),
    fServerRTPPort(serverRTPPort), fServerRTCPPort(serverRTCPPort),
    fRTPSink(rtpSink), fUDPSink(udpSink), fStreamDuration(master.duration()),
    fTotalBW(totalBW), fRTCPInstance(NULL) /* created later */,
    fMediaSource(mediaSource), fRTPgs(rtpGS), fRTCPgs(rtcpGS) {
}

StreamState::~StreamState() {
  reclaim();
}

void StreamState
::startPlaying(Destinations* dests,
           TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,
           ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,
           void* serverRequestAlternativeByteHandlerClientData) {
  if (dests == NULL) return;

  if (fRTCPInstance == NULL && fRTPSink != NULL) {
    // Create (and start) a 'RTCP instance' for this RTP sink:
    fRTCPInstance
      = RTCPInstance::createNew(fRTPSink->envir(), fRTCPgs,
                fTotalBW, (unsigned char*)fMaster.fCNAME,
                fRTPSink, NULL /* we're a server */);
        // Note: This starts RTCP running automatically
  }

  if (dests->isTCP) {
    // Change RTP and RTCP to use the TCP socket instead of UDP:
    if (fRTPSink != NULL) {
      fRTPSink->addStreamSocket(dests->tcpSocketNum, dests->rtpChannelId);
      fRTPSink->setServerRequestAlternativeByteHandler(dests->tcpSocketNum, serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);
    }
    if (fRTCPInstance != NULL) {
      fRTCPInstance->addStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId);
      fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId,
                      rtcpRRHandler, rtcpRRHandlerClientData);
    }
  } else {
    // Tell the RTP and RTCP 'groupsocks' about this destination
    // (in case they don't already have it):
    if (fRTPgs != NULL) fRTPgs->addDestination(dests->addr, dests->rtpPort);
    if (fRTCPgs != NULL) fRTCPgs->addDestination(dests->addr, dests->rtcpPort);
    if (fRTCPInstance != NULL) {
      fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,
                      rtcpRRHandler, rtcpRRHandlerClientData);
    }
  }

  if (!fAreCurrentlyPlaying && fMediaSource != NULL) {
    if (fRTPSink != NULL) {
      fRTPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
      fAreCurrentlyPlaying = True;
    } else if (fUDPSink != NULL) {
      fUDPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);
      fAreCurrentlyPlaying = True;
    }
  }
}

void StreamState::pause() {
  if (fRTPSink != NULL) fRTPSink->stopPlaying();
  if (fUDPSink != NULL) fUDPSink->stopPlaying();
  fAreCurrentlyPlaying = False;
}

void StreamState::endPlaying(Destinations* dests) {
  if (dests->isTCP) {
    if (fRTPSink != NULL) {
      fRTPSink->removeStreamSocket(dests->tcpSocketNum, dests->rtpChannelId);
    }
    if (fRTCPInstance != NULL) {
      fRTCPInstance->removeStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId);
      fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId,
                      NULL, NULL);
    }
  } else {
    // Tell the RTP and RTCP 'groupsocks' to stop using these destinations:
    if (fRTPgs != NULL) fRTPgs->removeDestination(dests->addr, dests->rtpPort);
    if (fRTCPgs != NULL) fRTCPgs->removeDestination(dests->addr, dests->rtcpPort);
    if (fRTCPInstance != NULL) {
      fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,
                      NULL, NULL);
    }
  }
}

void StreamState::reclaim() {
  // Delete allocated media objects
  Medium::close(fRTCPInstance) /* will send a RTCP BYE */; fRTCPInstance = NULL;
  Medium::close(fRTPSink); fRTPSink = NULL;
  Medium::close(fUDPSink); fUDPSink = NULL;

  fMaster.closeStreamSource(fMediaSource); fMediaSource = NULL;
  if (fMaster.fLastStreamToken == this) fMaster.fLastStreamToken = NULL;

  delete fRTPgs; fRTPgs = NULL;
  delete fRTCPgs; fRTCPgs = NULL;
}
