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• Problems with Streaming a Multicast RTSP Stream with Live555

  16 juin 2014, par ALM865

  I am having trouble setting up a Multicast RTSP session using Live555. The examples included with Live555 are mostly irrelevant as they deal with reading in files and my code differs because it reads in encoded frames generated from a FFMPEG thread within my own program (no pipes, no saving to disk, it is genuinely passing pointers to memory that contain the encoded frames for Live555 to stream).

  My Live555 project that uses a custom Server Media Subsession so that I can receive data from an FFMPEG thread within my program (instead of Live555’s default reading from a file, yuk !). This is a requirement of my program as it reads in a GigEVision stream in one thread, sends the decoded raw RGB packets to the FFMPEG thread, which then in turn sends the encoded frames off to Live555 for RTSP streaming.

  For the life of me I can’t work out how to send the RTSP stream as multicast instead of unicast !

  Just a note, my program works perfectly at the moment streaming Unicast, so there is nothing wrong with my Live555 implementation (before you go crazy picking out irrelevant errors !). I just need to know how to modify my existing code to stream Multicast instead of Unicast.

  My program is way too big to upload and share so I’m just going to share the important bits :

  Live_AnalysingServerMediaSubsession.h

  #ifndef _ANALYSING_SERVER_MEDIA_SUBSESSION_HH
  
  #define _ANALYSING_SERVER_MEDIA_SUBSESSION_HH
  
  
  
  #include 
  
  #include "Live_AnalyserInput.h"
  
  
  
  class AnalysingServerMediaSubsession: public OnDemandServerMediaSubsession {
  
  
  
  public:
  
    static AnalysingServerMediaSubsession*
  
    createNew(UsageEnvironment& env, AnalyserInput& analyserInput, unsigned estimatedBitrate,
  
          Boolean iFramesOnly = False,
  
              double vshPeriod = 5.0
  
              /* how often (in seconds) to inject a Video_Sequence_Header,
  
                 if one doesn't already appear in the stream */);
  
  
  
  protected: // we're a virtual base class
  
    AnalysingServerMediaSubsession(UsageEnvironment& env, AnalyserInput& AnalyserInput, unsigned estimatedBitrate, Boolean iFramesOnly, double vshPeriod);
  
    virtual ~AnalysingServerMediaSubsession();
  
  
  
  protected:
  
    AnalyserInput& fAnalyserInput;
  
    unsigned fEstimatedKbps;
  
  
  
  private:
  
    Boolean fIFramesOnly;
  
    double fVSHPeriod;
  
  
  
    // redefined virtual functions
  
    virtual FramedSource* createNewStreamSource(unsigned clientSessionId, unsigned& estBitrate);
  
    virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock, unsigned char rtpPayloadTypeIfDynamic, FramedSource* inputSource);
  
  
  
  };
  
  
  
  #endif

  And "Live_AnalysingServerMediaSubsession.cpp"

  #include "Live_AnalysingServerMediaSubsession.h"
  
  #include 
  
  #include 
  
  #include 
  
  
  
  AnalysingServerMediaSubsession* AnalysingServerMediaSubsession::createNew(UsageEnvironment& env, AnalyserInput& wisInput, unsigned estimatedBitrate,
  
      Boolean iFramesOnly,
  
      double vshPeriod) {
  
          return new AnalysingServerMediaSubsession(env, wisInput, estimatedBitrate,
  
              iFramesOnly, vshPeriod);
  
  }
  
  
  
  AnalysingServerMediaSubsession
  
      ::AnalysingServerMediaSubsession(UsageEnvironment& env, AnalyserInput& analyserInput,   unsigned estimatedBitrate, Boolean iFramesOnly, double vshPeriod) 
  
      : OnDemandServerMediaSubsession(env, True /*reuse the first source*/),
  
  
  
      fAnalyserInput(analyserInput), fIFramesOnly(iFramesOnly), fVSHPeriod(vshPeriod) {
  
          fEstimatedKbps = (estimatedBitrate + 500)/1000;
  
  
  
  }
  
  
  
  AnalysingServerMediaSubsession
  
      ::~AnalysingServerMediaSubsession() {
  
  }
  
  
  
  FramedSource* AnalysingServerMediaSubsession ::createNewStreamSource(unsigned /*clientSessionId*/, unsigned& estBitrate) {
  
      estBitrate = fEstimatedKbps;
  
  
  
      // Create a framer for the Video Elementary Stream:
  
      //LOG_MSG("Create Net Stream Source [%d]", estBitrate);
  
  
  
      return MPEG1or2VideoStreamDiscreteFramer::createNew(envir(), fAnalyserInput.videoSource());
  
  }
  
  
  
  RTPSink* AnalysingServerMediaSubsession ::createNewRTPSink(Groupsock* rtpGroupsock, unsigned char /*rtpPayloadTypeIfDynamic*/, FramedSource* /*inputSource*/) {
  
      setVideoRTPSinkBufferSize();
  
      /*
  
      struct in_addr destinationAddress;
  
      destinationAddress.s_addr = inet_addr("239.255.12.42");
  
  
  
      rtpGroupsock->addDestination(destinationAddress,8888);
  
      rtpGroupsock->multicastSendOnly();
  
      */
  
      return MPEG1or2VideoRTPSink::createNew(envir(), rtpGroupsock);
  
  }

  Live_AnalyserSouce.h

  #ifndef _ANALYSER_SOURCE_HH
  
  #define _ANALYSER_SOURCE_HH
  
  
  
  #ifndef _FRAMED_SOURCE_HH
  
  #include "FramedSource.hh"
  
  #endif
  
  
  
  class FFMPEG;
  
  
  
  // The following class can be used to define specific encoder parameters
  
  class AnalyserParameters {
  
  public:
  
    FFMPEG * Encoding_Source;
  
  };
  
  
  
  class AnalyserSource: public FramedSource {
  
  public:
  
    static AnalyserSource* createNew(UsageEnvironment& env, FFMPEG * E_Source);
  
    static unsigned GetRefCount();
  
  
  
  
  
  public:
  
    static EventTriggerId eventTriggerId;
  
  
  
  protected:
  
    AnalyserSource(UsageEnvironment& env, FFMPEG *  E_Source);
  
    // called only by createNew(), or by subclass constructors
  
    virtual ~AnalyserSource();
  
  
  
  private:
  
    // redefined virtual functions:
  
    virtual void doGetNextFrame();
  
  
  
  private:
  
    static void deliverFrame0(void* clientData);
  
    void deliverFrame();
  
  
  
  
  
  private:
  
    static unsigned referenceCount; // used to count how many instances of this class currently exist
  
    FFMPEG * Encoding_Source;
  
  
  
    unsigned int Last_Sent_Frame_ID;
  
  };
  
  
  
  #endif

  Live_AnalyserSource.cpp

  #include "Live_AnalyserSource.h"
  
  #include  // for "gettimeofday()"
  
  #include "FFMPEGClass.h"
  
  
  
  AnalyserSource* AnalyserSource::createNew(UsageEnvironment& env, FFMPEG * E_Source) {
  
    return new AnalyserSource(env, E_Source);
  
  }
  
  
  
  
  
  EventTriggerId AnalyserSource::eventTriggerId = 0;
  
  
  
  unsigned AnalyserSource::referenceCount = 0;
  
  
  
  AnalyserSource::AnalyserSource(UsageEnvironment& env, FFMPEG * E_Source) : FramedSource(env), Encoding_Source(E_Source) {
  
    if (referenceCount == 0) {
  
      // Any global initialization of the device would be done here:
  
  
  
    }
  
    ++referenceCount;
  
  
  
    // Any instance-specific initialization of the device would be done here:
  
    Last_Sent_Frame_ID = 0;
  
  
  
    /* register us with the Encoding thread so we'll get notices when new frame data turns up.. */
  
    Encoding_Source->RegisterRTSP_Source(&(env.taskScheduler()), this);
  
  
  
    // We arrange here for our "deliverFrame" member function to be called
  
    // whenever the next frame of data becomes available from the device.
  
    //
  
    // If the device can be accessed as a readable socket, then one easy way to do this is using a call to
  
    //     envir().taskScheduler().turnOnBackgroundReadHandling( ... )
  
    // (See examples of this call in the "liveMedia" directory.)
  
    //
  
    // If, however, the device *cannot* be accessed as a readable socket, then instead we can implement is using 'event triggers':
  
    // Create an 'event trigger' for this device (if it hasn't already been done):
  
    if (eventTriggerId == 0) {
  
      eventTriggerId = envir().taskScheduler().createEventTrigger(deliverFrame0);
  
    }
  
  }
  
  
  
  AnalyserSource::~AnalyserSource() {
  
    // Any instance-specific 'destruction' (i.e., resetting) of the device would be done here:
  
  
  
    /* de-register this source from the Encoding thread, since we no longer need notices.. */
  
    Encoding_Source->Un_RegisterRTSP_Source(this);
  
  
  
    --referenceCount;
  
    if (referenceCount == 0) {
  
      // Any global 'destruction' (i.e., resetting) of the device would be done here:
  
  
  
      // Reclaim our 'event trigger'
  
      envir().taskScheduler().deleteEventTrigger(eventTriggerId);
  
      eventTriggerId = 0;
  
    }
  
  
  
  }
  
  
  
  unsigned AnalyserSource::GetRefCount() {
  
    return referenceCount;
  
  }
  
  
  
  void AnalyserSource::doGetNextFrame() {
  
    // This function is called (by our 'downstream' object) when it asks for new data.
  
    //LOG_MSG("Do Next Frame..");
  
    // Note: If, for some reason, the source device stops being readable (e.g., it gets closed), then you do the following:
  
    //if (0 /* the source stops being readable */ /*%%% TO BE WRITTEN %%%*/) {
  
    unsigned int FrameID = Encoding_Source->GetFrameID();
  
    if (FrameID == 0){
  
      //LOG_MSG("No Data. Close");
  
      handleClosure(this);
  
      return;
  
    }
  
  
  
  
  
  
  
    // If a new frame of data is immediately available to be delivered, then do this now:
  
    if (Last_Sent_Frame_ID != FrameID){
  
      deliverFrame();
  
      //DEBUG_MSG("Frame ID: %d",FrameID);
  
    }
  
  
  
    // No new data is immediately available to be delivered.  We don't do anything more here.
  
    // Instead, our event trigger must be called (e.g., from a separate thread) when new data becomes available.
  
  }
  
  
  
  void AnalyserSource::deliverFrame0(void* clientData) {
  
    ((AnalyserSource*)clientData)->deliverFrame();
  
  }
  
  
  
  void AnalyserSource::deliverFrame() {
  
  
  
    if (!isCurrentlyAwaitingData()) return; // we're not ready for the data yet
  
  
  
  
  
    static u_int8_t* newFrameDataStart;
  
    static unsigned newFrameSize = 0;
  
  
  
    /* get the data frame from the Encoding thread.. */
  
    if (Encoding_Source->GetFrame(&newFrameDataStart, &newFrameSize, &Last_Sent_Frame_ID)){
  
      if (newFrameDataStart!=NULL) {
  
          /* This should never happen, but check anyway.. */
  
          if (newFrameSize > fMaxSize) {
  
            fFrameSize = fMaxSize;
  
            fNumTruncatedBytes = newFrameSize - fMaxSize;
  
          } else {
  
            fFrameSize = newFrameSize;
  
          }
  
          gettimeofday(&fPresentationTime, NULL); // If you have a more accurate time - e.g., from an encoder - then use that instead.
  
          // If the device is *not* a 'live source' (e.g., it comes instead from a file or buffer), then set "fDurationInMicroseconds" here.
  
          /* move the data to be sent off.. */
  
          memmove(fTo, newFrameDataStart, fFrameSize);
  
  
  
          /* release the Mutex we had on the Frame's buffer.. */
  
          Encoding_Source->ReleaseFrame();
  
      }
  
      else {
  
          //AM Added, something bad happened
  
          //ALTRACE("LIVE555: FRAME NULL\n");
  
          fFrameSize=0;
  
          fTo=NULL;
  
          handleClosure(this);
  
      }
  
    }
  
    else {
  
      //LOG_MSG("Closing Connection due to Frame Error..");
  
      handleClosure(this);
  
    }
  
  
  
  
  
    // After delivering the data, inform the reader that it is now available:
  
    FramedSource::afterGetting(this);
  
  }

  Live_AnalyserInput.cpp

  #include "Live_AnalyserInput.h"
  
  #include "Live_AnalyserSource.h"
  
  
  
  
  
  ////////// WISInput implementation //////////
  
  
  
  AnalyserInput* AnalyserInput::createNew(UsageEnvironment& env, FFMPEG *Encoder) {
  
    if (!fHaveInitialized) {
  
      //if (!initialize(env)) return NULL;
  
      fHaveInitialized = True;
  
    }
  
  
  
    return new AnalyserInput(env, Encoder);
  
  }
  
  
  
  
  
  FramedSource* AnalyserInput::videoSource() {
  
    if (fOurVideoSource == NULL || AnalyserSource::GetRefCount() == 0) {
  
      fOurVideoSource = AnalyserSource::createNew(envir(), m_Encoder);
  
    }
  
    return fOurVideoSource;
  
  }
  
  
  
  
  
  AnalyserInput::AnalyserInput(UsageEnvironment& env, FFMPEG *Encoder): Medium(env), m_Encoder(Encoder) {
  
  }
  
  
  
  AnalyserInput::~AnalyserInput() {
  
    /* When we get destroyed, make sure our source is also destroyed.. */
  
    if (fOurVideoSource != NULL && AnalyserSource::GetRefCount() != 0) {
  
      AnalyserSource::handleClosure(fOurVideoSource);
  
    }
  
  }
  
  
  
  
  
  
  
  
  
  Boolean AnalyserInput::fHaveInitialized = False;
  
  int AnalyserInput::fOurVideoFileNo = -1;
  
  FramedSource* AnalyserInput::fOurVideoSource = NULL;

  Live_AnalyserInput.h

  #ifndef _ANALYSER_INPUT_HH
  
  #define _ANALYSER_INPUT_HH
  
  
  
  #include 
  
  #include "FFMPEGClass.h"
  
  
  
  
  
  class AnalyserInput: public Medium {
  
  public:
  
    static AnalyserInput* createNew(UsageEnvironment& env, FFMPEG *Encoder);
  
  
  
    FramedSource* videoSource();
  
  
  
  private:
  
    AnalyserInput(UsageEnvironment& env, FFMPEG *Encoder); // called only by createNew()
  
    virtual ~AnalyserInput();
  
  
  
  private:
  
    friend class WISVideoOpenFileSource;
  
    static Boolean fHaveInitialized;
  
    static int fOurVideoFileNo;
  
    static FramedSource* fOurVideoSource;
  
    FFMPEG *m_Encoder;
  
  };
  
  
  
  // Functions to set the optimal buffer size for RTP sink objects.
  
  // These should be called before each RTPSink is created.
  
  #define VIDEO_MAX_FRAME_SIZE 300000
  
  inline void setVideoRTPSinkBufferSize() { OutPacketBuffer::maxSize = VIDEO_MAX_FRAME_SIZE; }
  
  
  
  #endif

  And finally the relevant code from my Live555 worker thread that starts the whole process :

      Stop_RTSP_Loop=0;
  
      //  MediaSession     *ms;
  
      TaskScheduler    *scheduler;
  
      UsageEnvironment *env ;
  
      //  RTSPClient       *rtsp;
  
      //  MediaSubsession  *Video_Sub;
  
  
  
      char RTSP_Address[1024];
  
      RTSP_Address[0]=0x00;
  
  
  
      if (m_Encoder == NULL){
  
          //DEBUG_MSG("No Video Encoder registered for the RTSP Encoder");
  
          return 0;
  
      }
  
  
  
      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:
  
  
  
      if (m_Enable_Pass){
  
          authDB = new UserAuthenticationDatabase;
  
          authDB->addUserRecord(UserN, PassW);
  
      }
  
      ////////// authDB = new UserAuthenticationDatabase;
  
      ////////// authDB->addUserRecord((char*)"Admin", (char*)"Admin"); // 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, 554, authDB);
  
      ServerMediaSession* sms;
  
  
  
      AnalyserInput* inputDevice;
  
  
  
  
  
      if (rtspServer == NULL) {
  
          TRACE("LIVE555: Failed to create RTSP server: %s\n", env->getResultMsg());
  
          return 0;
  
      }
  
      else {
  
          char const* descriptionString = "Session streamed by \"IMC Server\"";
  
  
  
  
  
  
  
          // Initialize the WIS input device:
  
          inputDevice = AnalyserInput::createNew(*env, m_Encoder);
  
          if (inputDevice == NULL) {
  
              TRACE("Live555: Failed to create WIS input device\n");
  
              return 0;
  
          }
  
          else {
  
              // A MPEG-1 or 2 video elementary stream:
  
              /* Increase the buffer size so we can handle the high res stream.. */
  
              OutPacketBuffer::maxSize = 300000;
  
              // NOTE: This *must* be a Video Elementary Stream; not a Program Stream
  
              sms = ServerMediaSession::createNew(*env, RTSP_Address, RTSP_Address, descriptionString);
  
  
  
              //sms->addSubsession(MPEG1or2VideoFileServerMediaSubsession::createNew(*env, inputFileName, reuseFirstSource, iFramesOnly));
  
  
  
              sms->addSubsession(AnalysingServerMediaSubsession::createNew(*env, *inputDevice, m_Encoder->Get_Bitrate()));
  
              //sms->addSubsession(WISMPEG1or2VideoServerMediaSubsession::createNew(sms->envir(), inputDevice, videoBitrate));
  
  
  
              rtspServer->addServerMediaSession(sms);
  
  
  
              //announceStream(rtspServer, sms, streamName, inputFileName);
  
              //LOG_MSG("Play this stream using the URL %s", rtspServer->rtspURL(sms));
  
  
  
          }
  
      }
  
  
  
      Stop_RTSP_Loop=0;
  
  
  
      for (;;)
  
      {
  
          /* The actual work is all carried out inside the LIVE555 Task scheduler */
  
          env->taskScheduler().doEventLoop(&amp;Stop_RTSP_Loop); // does not return
  
  
  
          if (mStop) {
  
              break;
  
          }
  
      }
  
  
  
      Medium::close(rtspServer); // will also reclaim "sms" and its "ServerMediaSubsession"s
  
      Medium::close(inputDevice);
  
  </password></username>

• FFMPEG Streaming using RTMP

  20 février 2014, par desto

  I'm trying to create a stream using ffmpeg to send a video to a Red5 Server. I've already managed to do this using this command :

  ffmpeg -re -y -i "Videos\Video1.mp4" -c:v libx264 -b:v 600k -r 25 -s 640x360 -t 40 -vf yadif -b:a 64k -ac 1 -ar 44100 -f flv "rtmp://192.168.0.12/live/videostream"

  My problem is, when ffmpeg finishes encoding the video, it stops the stream, and thus cuts the video short for 5-10 seconds (for short videos), but this gets worse on larger videos.

  Is there a way to stop this behavior ?
  I was trying to add a blank 10 second video before and after the original video, but due to some encoding options, I always end up losing audio. And this only kind-of works on the short videos, but on longer videos the problem is still there.

  Any recommendations ?

• RTSP streaming on Android client using FFMpeg

  10 août 2013, par rurtle

  I am working on a hobby project the goal for which is to develop an Android application capable of streaming live feeds captured through web cams in a LAN setting using FFMpeg as the underlying engine. So far, I did the following -

  A. Compiling and generating FFMpeg related libraries for the following releases -

  FFMpeg version : 2.0
  NDK version : r8e & r9
  Android Platform version : android-16 & android-18thisthisthisthis
  Toolchain version : 4.6 & 4.8
  Platform built on : Fedora 18 (x86_64)
  

  B. Creating the files Android.mk & Application.mk in appropriate path.

  However, when it came to writing the native code for accessing appropriate functionality of FFMpeg from the application layer using Java, I'm stuck with following questions -

  a) Which all of FFMpeg's features I need to make available from native to app layer for streaming real-time feeds ?
  b) In order to compile FFMpeg for Android, I followed this link. Whether the compilation options are sufficient for handling *.sdp streams or do I need to modify it ?
  c) Do I need to make use of live555 ?
  

  I am totally new to FFMpeg and Android application development and this is going to be my first serious project for Android platform. I have been searching for relevant tutorials dealing with RTSP streaming using FFMpeg for a while now without much success. Moreover, I tried the latest development build of VLC player and found it to be great for streaming real-time feeds. However, it's a complex beast and the goal for my project is of quite limited nature, mostly learning - in a short time span.

  Could you suggest some pointers (e.g. links, documents or sample code) on how can I write the native code for utilizing FFMpeg library and subsequently use those functionality from the app layer for streaming real-time feeds ? Moreover, will really appreciate if you could let me know the kind of background knowledge necessary for this project from a functional standpoint (in a language agnostic sense).