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• FFmpeg does not decode h264 stream

  5 juillet 2012, par HAPPY_TIGER

  I am trying to decode h264 stream from rtsp server and render it on iPhone.

  I found some libraries and read some articles about it.

  Libraries are from dropCam for iPhone called RTSPClient and DecoderWrapper.

  But I can not decode frame data with DecodeWrapper that using on ffmpeg.

  Here are my code.

  VideoViewer.m
  
  
  
  - (void)didReceiveFrame:(NSData*)frameData presentationTime:(NSDate*)presentationTime
  
  {
  
      [VideoDecoder staticInitialize];
  
      mConverter = [[VideoDecoder alloc] initWithCodec:kVCT_H264 colorSpace:kVCS_RGBA32 width:320 height:240 privateData:nil];
  
  
  
  
  
      [mConverter decodeFrame:frameData];
  
  
  
      if ([mConverter isFrameReady]) {
  
          UIImage *imageData =[mConverter getDecodedFrame];
  
          if (imageData) {
  
              [mVideoView setImage:imageData];
  
              NSLog(@"decoded!");
  
          }
  
      }
  
  }
  
  
  
  ---VideoDecoder.m---
  
  - (id)initWithCodec:(enum VideoCodecType)codecType 
  
           colorSpace:(enum VideoColorSpace)colorSpace 
  
                width:(int)width 
  
               height:(int)height 
  
          privateData:(NSData*)privateData {
  
      if(self = [super init]) {
  
  
  
          codec = avcodec_find_decoder(CODEC_ID_H264);
  
          codecCtx = avcodec_alloc_context();
  
  
  
          // Note: for H.264 RTSP streams, the width and height are usually not specified (width and height are 0).  
  
          // These fields will become filled in once the first frame is decoded and the SPS is processed.
  
          codecCtx->width = width;
  
          codecCtx->height = height;
  
  
  
          codecCtx->extradata = av_malloc([privateData length]);
  
          codecCtx->extradata_size = [privateData length];
  
          [privateData getBytes:codecCtx->extradata length:codecCtx->extradata_size];
  
          codecCtx->pix_fmt = PIX_FMT_RGBA;
  
  #ifdef SHOW_DEBUG_MV
  
          codecCtx->debug_mv = 0xFF;
  
  #endif
  
  
  
          srcFrame = avcodec_alloc_frame();
  
          dstFrame = avcodec_alloc_frame();
  
  
  
          int res = avcodec_open(codecCtx, codec);
  
          if (res < 0)
  
          {
  
              NSLog(@"Failed to initialize decoder");
  
          }
  
  
  
      }
  
  
  
      return self;    
  
  }
  
  
  
  - (void)decodeFrame:(NSData*)frameData {
  
  
  
  
  
      AVPacket packet = {0};
  
      packet.data = (uint8_t*)[frameData bytes];
  
      packet.size = [frameData length];
  
  
  
      int frameFinished=0;
  
      NSLog(@"Packet size===>%d",packet.size);
  
      // Is this a packet from the video stream?
  
      if(packet.stream_index==0)
  
      {
  
          int res = avcodec_decode_video2(codecCtx, srcFrame, &frameFinished, &packet);
  
          NSLog(@"Res value===>%d",res);
  
          NSLog(@"frame data===>%d",(int)srcFrame->data);
  
          if (res < 0)
  
          {
  
              NSLog(@"Failed to decode frame");
  
          }
  
      }
  
      else 
  
      {
  
          NSLog(@"No video stream found");
  
      }
  
  
  
  
  
      // Need to delay initializing the output buffers because we don't know the dimensions until we decode the first frame.
  
      if (!outputInit) {
  
          if (codecCtx->width > 0 && codecCtx->height > 0) {
  
  #ifdef _DEBUG
  
              NSLog(@"Initializing decoder with frame size of: %dx%d", codecCtx->width, codecCtx->height);
  
  #endif
  
  
  
              outputBufLen = avpicture_get_size(PIX_FMT_RGBA, codecCtx->width, codecCtx->height);
  
              outputBuf = av_malloc(outputBufLen);
  
  
  
              avpicture_fill((AVPicture*)dstFrame, outputBuf, PIX_FMT_RGBA, codecCtx->width, codecCtx->height);
  
  
  
              convertCtx = sws_getContext(codecCtx->width, codecCtx->height, codecCtx->pix_fmt,  codecCtx->width, 
  
                                          codecCtx->height, PIX_FMT_RGBA, SWS_FAST_BILINEAR, NULL, NULL, NULL); 
  
  
  
              outputInit = YES;
  
              frameFinished=1;
  
          }
  
          else {
  
              NSLog(@"Could not get video output dimensions");
  
          }
  
      }
  
  
  
      if (frameFinished)
  
          frameReady = YES;
  
  
  
  }

  The console shows me as follows.

  2011-05-16 20:16:04.223 RTSPTest1[41226:207] Packet size===>359
  
  [h264 @ 0x5815c00] no frame!
  
  2011-05-16 20:16:04.223 RTSPTest1[41226:207] Res value===>-1
  
  2011-05-16 20:16:04.224 RTSPTest1[41226:207] frame data===>101791200
  
  2011-05-16 20:16:04.224 RTSPTest1[41226:207] Failed to decode frame
  
  2011-05-16 20:16:04.225 RTSPTest1[41226:207] decoded!
  
  2011-05-16 20:16:04.226 RTSPTest1[41226:207] Packet size===>424
  
  [h264 @ 0x5017c00] no frame!
  
  2011-05-16 20:16:04.226 RTSPTest1[41226:207] Res value===>-1
  
  2011-05-16 20:16:04.227 RTSPTest1[41226:207] frame data===>81002704
  
  2011-05-16 20:16:04.227 RTSPTest1[41226:207] Failed to decode frame
  
  2011-05-16 20:16:04.228 RTSPTest1[41226:207] decoded!
  
  2011-05-16 20:16:04.229 RTSPTest1[41226:207] Packet size===>424
  
  [h264 @ 0x581d000] no frame!
  
  2011-05-16 20:16:04.229 RTSPTest1[41226:207] Res value===>-1
  
  2011-05-16 20:16:04.230 RTSPTest1[41226:207] frame data===>101791616
  
  2011-05-16 20:16:04.230 RTSPTest1[41226:207] Failed to decode frame
  
  2011-05-16 20:16:04.231 RTSPTest1[41226:207] decoded!
  
  . . . .  .

  But the simulator shows nothing.

  What's wrong with my code.

  Help me solve this problem.

  Thanks for your answers.

• Live555 : X264 Stream Live source based on "testOnDemandRTSPServer"

  12 janvier 2017, par user2660369

  I am trying to create a rtsp Server that streams the OpenGL output of my program. I had a look at How to write a Live555 FramedSource to allow me to stream H.264 live, but I need the stream to be unicast. So I had a look at testOnDemandRTSPServer. Using the same Code fails. To my understanding I need to provide memory in which I store my h264 frames so the OnDemandServer can read them on Demand.

  H264VideoStreamServerMediaSubsession.cpp

  H264VideoStreamServerMediaSubsession*
  
  H264VideoStreamServerMediaSubsession::createNew(UsageEnvironment& env,
  
                            Boolean reuseFirstSource) {
  
    return new H264VideoStreamServerMediaSubsession(env, reuseFirstSource);
  
  }
  
  
  
  H264VideoStreamServerMediaSubsession::H264VideoStreamServerMediaSubsession(UsageEnvironment& env, Boolean reuseFirstSource)
  
    : OnDemandServerMediaSubsession(env, reuseFirstSource), fAuxSDPLine(NULL), fDoneFlag(0), fDummyRTPSink(NULL) {
  
  }
  
  
  
  H264VideoStreamServerMediaSubsession::~H264VideoStreamServerMediaSubsession() {
  
    delete[] fAuxSDPLine;
  
  }
  
  
  
  static void afterPlayingDummy(void* clientData) {
  
    H264VideoStreamServerMediaSubsession* subsess = (H264VideoStreamServerMediaSubsession*)clientData;
  
    subsess->afterPlayingDummy1();
  
  }
  
  
  
  void H264VideoStreamServerMediaSubsession::afterPlayingDummy1() {
  
    // Unschedule any pending 'checking' task:
  
    envir().taskScheduler().unscheduleDelayedTask(nextTask());
  
    // Signal the event loop that we're done:
  
    setDoneFlag();
  
  }
  
  
  
  static void checkForAuxSDPLine(void* clientData) {
  
    H264VideoStreamServerMediaSubsession* subsess = (H264VideoStreamServerMediaSubsession*)clientData;
  
    subsess->checkForAuxSDPLine1();
  
  }
  
  
  
  void H264VideoStreamServerMediaSubsession::checkForAuxSDPLine1() {
  
    char const* dasl;
  
  
  
    if (fAuxSDPLine != NULL) {
  
      // Signal the event loop that we're done:
  
      setDoneFlag();
  
    } else if (fDummyRTPSink != NULL && (dasl = fDummyRTPSink->auxSDPLine()) != NULL) {
  
      fAuxSDPLine = strDup(dasl);
  
      fDummyRTPSink = NULL;
  
  
  
      // Signal the event loop that we're done:
  
      setDoneFlag();
  
    } else {
  
      // try again after a brief delay:
  
      int uSecsToDelay = 100000; // 100 ms
  
      nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecsToDelay,
  
                    (TaskFunc*)checkForAuxSDPLine, this);
  
    }
  
  }
  
  
  
  char const* H264VideoStreamServerMediaSubsession::getAuxSDPLine(RTPSink* rtpSink, FramedSource* inputSource) {
  
    if (fAuxSDPLine != NULL) return fAuxSDPLine; // it's already been set up (for a previous client)
  
  
  
    if (fDummyRTPSink == NULL) { // we're not already setting it up for another, concurrent stream
  
      // Note: For H264 video files, the 'config' information ("profile-level-id" and "sprop-parameter-sets") isn't known
  
      // until we start reading the file.  This means that "rtpSink"s "auxSDPLine()" will be NULL initially,
  
      // and we need to start reading data from our file until this changes.
  
      fDummyRTPSink = rtpSink;
  
  
  
      // Start reading the file:
  
      fDummyRTPSink->startPlaying(*inputSource, afterPlayingDummy, this);
  
  
  
      // Check whether the sink's 'auxSDPLine()' is ready:
  
      checkForAuxSDPLine(this);
  
    }
  
  
  
    envir().taskScheduler().doEventLoop(&fDoneFlag);
  
  
  
    return fAuxSDPLine;
  
  }
  
  
  
  FramedSource* H264VideoStreamServerMediaSubsession::createNewStreamSource(unsigned /*clientSessionId*/, unsigned& estBitrate) {
  
    estBitrate = 500; // kb
  
    megamol::remotecontrol::View3D_MRC *parent = (megamol::remotecontrol::View3D_MRC*)this->parent;
  
    return H264VideoStreamFramer::createNew(envir(), parent->h264FramedSource);
  
  }
  
  
  
  RTPSink* H264VideoStreamServerMediaSubsession::createNewRTPSink(Groupsock* rtpGroupsock, unsigned char rtpPayloadTypeIfDynamic, FramedSource* /*inputSource*/) {
  
    return H264VideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic);
  
  }

  FramedSource.cpp

  H264FramedSource* H264FramedSource::createNew(UsageEnvironment& env,
  
                                            unsigned preferredFrameSize,
  
                                            unsigned playTimePerFrame)
  
  {
  
      return new H264FramedSource(env, preferredFrameSize, playTimePerFrame);
  
  }
  
  
  
  H264FramedSource::H264FramedSource(UsageEnvironment& env,
  
                                 unsigned preferredFrameSize,
  
                                 unsigned playTimePerFrame)
  
      : FramedSource(env),
  
      fPreferredFrameSize(fMaxSize),
  
      fPlayTimePerFrame(playTimePerFrame),
  
      fLastPlayTime(0),
  
      fCurIndex(0)
  
  {
  
  
  
      x264_param_default_preset(&param, "veryfast", "zerolatency");
  
      param.i_threads = 1;
  
      param.i_width = 1024;
  
      param.i_height = 768;
  
      param.i_fps_num = 30;
  
      param.i_fps_den = 1;
  
      // Intra refres:
  
      param.i_keyint_max = 60;
  
      param.b_intra_refresh = 1;
  
      //Rate control:
  
      param.rc.i_rc_method = X264_RC_CRF;
  
      param.rc.f_rf_constant = 25;
  
      param.rc.f_rf_constant_max = 35;
  
      param.i_sps_id = 7;
  
      //For streaming:
  
      param.b_repeat_headers = 1;
  
      param.b_annexb = 1;
  
      x264_param_apply_profile(&param, "baseline");
  
  
  
      param.i_log_level = X264_LOG_ERROR;
  
  
  
      encoder = x264_encoder_open(&param);
  
      pic_in.i_type            = X264_TYPE_AUTO;
  
      pic_in.i_qpplus1         = 0;
  
      pic_in.img.i_csp         = X264_CSP_I420;
  
      pic_in.img.i_plane       = 3;
  
  
  
  
  
      x264_picture_alloc(&pic_in, X264_CSP_I420, 1024, 768);
  
  
  
      convertCtx = sws_getContext(1024, 768, PIX_FMT_RGBA, 1024, 768, PIX_FMT_YUV420P, SWS_FAST_BILINEAR, NULL, NULL, NULL);
  
      eventTriggerId = envir().taskScheduler().createEventTrigger(deliverFrame0);
  
  }
  
  
  
  H264FramedSource::~H264FramedSource()
  
  {
  
      envir().taskScheduler().deleteEventTrigger(eventTriggerId);
  
      eventTriggerId = 0;
  
  }
  
  
  
  void H264FramedSource::AddToBuffer(uint8_t* buf, int surfaceSizeInBytes)
  
  {
  
      uint8_t* surfaceData = (new uint8_t[surfaceSizeInBytes]);
  
  
  
      memcpy(surfaceData, buf, surfaceSizeInBytes);
  
  
  
      int srcstride = 1024*4;
  
      sws_scale(convertCtx, &surfaceData, &srcstride,0, 768, pic_in.img.plane, pic_in.img.i_stride);
  
      x264_nal_t* nals = NULL;
  
      int i_nals = 0;
  
      int frame_size = -1;
  
  
  
  
  
      frame_size = x264_encoder_encode(encoder, &nals, &i_nals, &pic_in, &pic_out);
  
  
  
      static bool finished = false;
  
  
  
      if (frame_size >= 0)
  
      {
  
      static bool alreadydone = false;
  
      if(!alreadydone)
  
      {
  
  
  
          x264_encoder_headers(encoder, &nals, &i_nals);
  
          alreadydone = true;
  
      }
  
      for(int i = 0; i < i_nals; ++i)
  
      {
  
          m_queue.push(nals[i]);
  
      }
  
      }
  
      delete [] surfaceData;
  
      surfaceData = nullptr;
  
  
  
      envir().taskScheduler().triggerEvent(eventTriggerId, this);
  
  }
  
  
  
  void H264FramedSource::doGetNextFrame()
  
  {
  
      deliverFrame();
  
  }
  
  
  
  void H264FramedSource::deliverFrame0(void* clientData)
  
  {
  
      ((H264FramedSource*)clientData)->deliverFrame();
  
  }
  
  
  
  void H264FramedSource::deliverFrame()
  
  {
  
      x264_nal_t nalToDeliver;
  
  
  
      if (fPlayTimePerFrame > 0 && fPreferredFrameSize > 0) {
  
      if (fPresentationTime.tv_sec == 0 && fPresentationTime.tv_usec == 0) {
  
          // This is the first frame, so use the current time:
  
          gettimeofday(&fPresentationTime, NULL);
  
      } else {
  
          // Increment by the play time of the previous data:
  
          unsigned uSeconds   = fPresentationTime.tv_usec + fLastPlayTime;
  
          fPresentationTime.tv_sec += uSeconds/1000000;
  
          fPresentationTime.tv_usec = uSeconds%1000000;
  
      }
  
  
  
      // Remember the play time of this data:
  
      fLastPlayTime = (fPlayTimePerFrame*fFrameSize)/fPreferredFrameSize;
  
      fDurationInMicroseconds = fLastPlayTime;
  
      } else {
  
      // We don't know a specific play time duration for this data,
  
      // so just record the current time as being the 'presentation time':
  
      gettimeofday(&fPresentationTime, NULL);
  
      }
  
  
  
      if(!m_queue.empty())
  
      {
  
      m_queue.wait_and_pop(nalToDeliver);
  
  
  
      uint8_t* newFrameDataStart = (uint8_t*)0xD15EA5E;
  
  
  
      newFrameDataStart = (uint8_t*)(nalToDeliver.p_payload);
  
      unsigned newFrameSize = nalToDeliver.i_payload;
  
  
  
      // Deliver the data here:
  
      if (newFrameSize > fMaxSize) {
  
          fFrameSize = fMaxSize;
  
          fNumTruncatedBytes = newFrameSize - fMaxSize;
  
      }
  
      else {
  
          fFrameSize = newFrameSize;
  
      }
  
  
  
      memcpy(fTo, nalToDeliver.p_payload, nalToDeliver.i_payload);
  
  
  
      FramedSource::afterGetting(this);
  
      }
  
  }

  Relevant part of the RTSP-Server Therad

    RTSPServer* rtspServer = RTSPServer::createNew(*(parent->env), 8554, NULL);
  
    if (rtspServer == NULL) {
  
      *(parent->env) << "Failed to create RTSP server: " << (parent->env)->getResultMsg() << "\n";
  
      exit(1);
  
    }
  
    char const* streamName = "Stream";
  
    parent->h264FramedSource = H264FramedSource::createNew(*(parent->env), 0, 0);
  
    H264VideoStreamServerMediaSubsession *h264VideoStreamServerMediaSubsession = H264VideoStreamServerMediaSubsession::createNew(*(parent->env), true);
  
    h264VideoStreamServerMediaSubsession->parent = parent;
  
    sms->addSubsession(h264VideoStreamServerMediaSubsession);
  
    rtspServer->addServerMediaSession(sms);
  
  
  
    parent->env->taskScheduler().doEventLoop(); // does not return

  Once a connection exists the render loop calls

  h264FramedSource->AddToBuffer(videoData, 1024*768*4);

• Unable to stream file onto localhost - ffmpeg

  18 octobre 2013, par trueblue

  I am new to ffmpeg/ffserver. I am trying to stream a local file named Trial onto a localhost using ffserver. I want to run the file in browser as http://localhost:8090/feed1.ffm
  I am executing the below command in Ubuntu(Trial is a Mpeg TS file) :

    ffmpeg -i Trial http://localhost:8090/feed1.ffm

  Upon execution of above command I am getting below error :

  FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.3, Copyright (c) 2000-2009 Fabrice Bellard, et al.
  
    configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.3 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static
  
    libavutil     49.15. 0 / 49.15. 0
  
    libavcodec    52.20. 1 / 52.20. 1
  
    libavformat   52.31. 0 / 52.31. 0
  
    libavdevice   52. 1. 0 / 52. 1. 0
  
    libavfilter    0. 4. 0 /  0. 4. 0
  
    libswscale     0. 7. 1 /  0. 7. 1
  
    libpostproc   51. 2. 0 / 51. 2. 0
  
    built on Jan 24 2013 19:42:59, gcc: 4.4.3
  
  
  
  Seems stream 0 codec frame rate differs from container frame rate: 119.88 (120000/1001) -> 59.94 (60000/1001)
  
  Input #0, mpegts, from 'Trial':
  
    Duration: 00:00:04.22, start: 0.177633, bitrate: 40368 kb/s
  
    Program 2 
  
      Stream #0.0[0x21]: Video: mpeg2video, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 45000 kb/s, 59.94 tbr, 90k tbn, 119.88 tbc
  
  Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
  
      Stream #0.0: Video: flv, yuv420p, 352x288, q=1-5, 100 kb/s, 1000k tbn, 15 tbc
  
      Stream #0.1: Audio: mp2, 44100 Hz, mono, s16, 32 kb/s
  
      Stream #0.2: Video: mpeg1video, yuv420p, 160x128, q=3-31, 64 kb/s, 1000k tbn, 3 tbc
  
      Stream #0.3: Audio: mp2, 22050 Hz, mono, s16, 64 kb/s
  
      Stream #0.4: Video: msmpeg4, yuv420p, 352x240, q=3-31, 256 kb/s, 1000k tbn, 15 tbc
  
  Could not find input stream matching output stream #0.1

  My ffserver.conf file goes like this :

  # Port on which the server is listening. You must select a different
  
  # port from your standard HTTP web server if it is running on the same
  
  # computer.
  
  Port 8090
  
  
  
  # Address on which the server is bound. Only useful if you have
  
  # several network interfaces.
  
  BindAddress 0.0.0.0
  
  
  
  # Number of simultaneous HTTP connections that can be handled. It has
  
  # to be defined *before* the MaxClients parameter, since it defines the
  
  # MaxClients maximum limit.
  
  MaxHTTPConnections 2000
  
  
  
  # Number of simultaneous requests that can be handled. Since FFServer
  
  # is very fast, it is more likely that you will want to leave this high
  
  # and use MaxBandwidth, below.
  
  MaxClients 1000
  
  
  
  # This the maximum amount of kbit/sec that you are prepared to
  
  # consume when streaming to clients.
  
  MaxBandwidth 1000
  
  
  
  # Access log file (uses standard Apache log file format)
  
  # '-' is the standard output.
  
  CustomLog -
  
  
  
  # Suppress that if you want to launch ffserver as a daemon.
  
  NoDaemon
  
  
  
  
  
  ##################################################################
  
  # Definition of the live feeds. Each live feed contains one video
  
  # and/or audio sequence coming from an ffmpeg encoder or another
  
  # ffserver. This sequence may be encoded simultaneously with several
  
  # codecs at several resolutions.
  
  
  
  <feed>
  
  
  
  # You must use &#39;ffmpeg&#39; to send a live feed to ffserver. In this
  
  # example, you can type:
  
  #
  
  # ffmpeg http://localhost:8090/feed1.ffm
  
  
  
  # ffserver can also do time shifting. It means that it can stream any
  
  # previously recorded live stream. The request should contain:
  
  # "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
  
  # a path where the feed is stored on disk. You also specify the
  
  # maximum size of the feed, where zero means unlimited. Default:
  
  # File=/tmp/feed_name.ffm FileMaxSize=5M
  
  File /tmp/feed1.ffm
  
  FileMaxSize 5M
  
  
  
  # You could specify
  
  # ReadOnlyFile /saved/specialvideo.ffm
  
  # This marks the file as readonly and it will not be deleted or updated.
  
  
  
  # Specify launch in order to start ffmpeg automatically.
  
  # First ffmpeg must be defined with an appropriate path if needed,
  
  # after that options can follow, but avoid adding the http:// field
  
  #Launch ffmpeg
  
  
  
  # Only allow connections from localhost to the feed.
  
  ACL allow 127.0.0.1
  
  
  
  </feed>
  
  
  
  
  
  
  
  <stream>
  
  Feed feed1.ffm
  
  Format swf
  
  VideoCodec flv
  
  VideoFrameRate 15
  
  VideoBufferSize 80000
  
  VideoBitRate 100
  
  VideoQMin 1
  
  VideoQMax 5
  
  VideoSize 352x288
  
  PreRoll 0
  
  Noaudio
  
  </stream>
  
  
  
  ##################################################################
  
  # Now you can define each stream which will be generated from the
  
  # original audio and video stream. Each format has a filename (here
  
  # &#39;test1.mpg&#39;). FFServer will send this stream when answering a
  
  # request containing this filename.
  
  
  
  <stream>
  
  
  
  # coming from live feed &#39;feed1&#39;
  
  Feed feed1.ffm
  
  
  
  # Format of the stream : you can choose among:
  
  # mpeg       : MPEG-1 multiplexed video and audio
  
  # mpegvideo  : only MPEG-1 video
  
  # mp2        : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
  
  # ogg        : Ogg format (Vorbis audio codec)
  
  # rm         : RealNetworks-compatible stream. Multiplexed audio and video.
  
  # ra         : RealNetworks-compatible stream. Audio only.
  
  # mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
  
  # jpeg       : Generate a single JPEG image.
  
  # asf        : ASF compatible streaming (Windows Media Player format).
  
  # swf        : Macromedia Flash compatible stream
  
  # avi        : AVI format (MPEG-4 video, MPEG audio sound)
  
  Format mpeg
  
  
  
  # Bitrate for the audio stream. Codecs usually support only a few
  
  # different bitrates.
  
  AudioBitRate 32
  
  
  
  # Number of audio channels: 1 = mono, 2 = stereo
  
  AudioChannels 1
  
  
  
  # Sampling frequency for audio. When using low bitrates, you should
  
  # lower this frequency to 22050 or 11025. The supported frequencies
  
  # depend on the selected audio codec.
  
  AudioSampleRate 44100
  
  
  
  # Bitrate for the video stream
  
  VideoBitRate 64
  
  
  
  
  
  # Ratecontrol buffer size
  
  VideoBufferSize 40
  
  
  
  # Number of frames per second
  
  VideoFrameRate 3
  
  
  
  # Size of the video frame: WxH (default: 160x128)
  
  # The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
  
  # qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
  
  # wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
  
  # hd1080
  
  VideoSize 160x128
  
  
  
  # Transmit only intra frames (useful for low bitrates, but kills frame rate).
  
  #VideoIntraOnly
  
  
  
  # If non-intra only, an intra frame is transmitted every VideoGopSize
  
  # frames. Video synchronization can only begin at an intra frame.
  
  VideoGopSize 12
  
  
  
  # More MPEG-4 parameters
  
  # VideoHighQuality
  
  # Video4MotionVector
  
  
  
  # Choose your codecs:
  
  #AudioCodec mp2
  
  #VideoCodec mpeg1video
  
  
  
  # Suppress audio
  
  #NoAudio
  
  
  
  # Suppress video
  
  #NoVideo
  
  
  
  #VideoQMin 3
  
  #VideoQMax 31
  
  
  
  # Set this to the number of seconds backwards in time to start. Note that
  
  # most players will buffer 5-10 seconds of video, and also you need to allow
  
  # for a keyframe to appear in the data stream.
  
  #Preroll 15
  
  
  
  # ACL:
  
  
  
  # You can allow ranges of addresses (or single addresses)
  
  #ACL ALLOW <first address="address"> <last address="address">
  
  
  
  # You can deny ranges of addresses (or single addresses)
  
  #ACL DENY <first address="address"> <last address="address">
  
  
  
  # You can repeat the ACL allow/deny as often as you like. It is on a per
  
  # stream basis. The first match defines the action. If there are no matches,
  
  # then the default is the inverse of the last ACL statement.
  
  #
  
  # Thus &#39;ACL allow localhost&#39; only allows access from localhost.
  
  # &#39;ACL deny 1.0.0.0 1.255.255.255&#39; would deny the whole of network 1 and
  
  # allow everybody else.
  
  
  
  </last></first></last></first></stream>
  
  
  
  
  
  ##################################################################
  
  # Example streams
  
  
  
  
  
  # Multipart JPEG
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format mpjpeg
  
  #VideoFrameRate 2
  
  #VideoIntraOnly
  
  #NoAudio
  
  #Strict -1
  
  #</stream>
  
  
  
  
  
  # Single JPEG
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format jpeg
  
  #VideoFrameRate 2
  
  #VideoIntraOnly
  
  ##VideoSize 352x240
  
  #NoAudio
  
  #Strict -1
  
  #</stream>
  
  
  
  
  
  
  
  # Flash
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format swf
  
  #VideoFrameRate 2
  
  #VideoIntraOnly
  
  #NoAudio
  
  #</stream>
  
  
  
  
  
  # ASF compatible
  
  
  
  <stream>
  
  Feed feed1.ffm
  
  Format asf
  
  VideoFrameRate 15
  
  VideoSize 352x240
  
  VideoBitRate 256
  
  VideoBufferSize 40
  
  VideoGopSize 30
  
  AudioBitRate 64
  
  StartSendOnKey
  
  </stream>
  
  
  
  
  
  # MP3 audio
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format mp2
  
  #AudioCodec mp3
  
  #AudioBitRate 64
  
  #AudioChannels 1
  
  #AudioSampleRate 44100
  
  #NoVideo
  
  #</stream>
  
  
  
  
  
  # Ogg Vorbis audio
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Title "Stream title"
  
  #AudioBitRate 64
  
  #AudioChannels 2
  
  #AudioSampleRate 44100
  
  #NoVideo
  
  #</stream>
  
  
  
  
  
  # Real with audio only at 32 kbits
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format rm
  
  #AudioBitRate 32
  
  #NoVideo
  
  #NoAudio
  
  #</stream>
  
  
  
  
  
  # Real with audio and video at 64 kbits
  
  
  
  #<stream>
  
  #Feed feed1.ffm
  
  #Format rm
  
  #AudioBitRate 32
  
  #VideoBitRate 128
  
  #VideoFrameRate 25
  
  #VideoGopSize 25
  
  #NoAudio
  
  #</stream>
  
  
  
  
  
  ##################################################################
  
  # A stream coming from a file: you only need to set the input
  
  # filename and optionally a new format. Supported conversions:
  
  #    AVI -> ASF
  
  
  
  #<stream>
  
  #File "/usr/local/httpd/htdocs/tlive.rm"
  
  #NoAudio
  
  #</stream>
  
  
  
  #<stream>
  
  #File "/usr/local/httpd/htdocs/test.asf"
  
  #NoAudio
  
  #Author "Me"
  
  #Copyright "Super MegaCorp"
  
  #Title "Test stream from disk"
  
  #Comment "Test comment"
  
  #</stream>
  
  
  
  
  
  ##################################################################
  
  # RTSP examples
  
  #
  
  # You can access this stream with the RTSP URL:
  
  #   rtsp://localhost:5454/test1-rtsp.mpg
  
  #
  
  # A non-standard RTSP redirector is also created. Its URL is:
  
  #   http://localhost:8090/test1-rtsp.rtsp
  
  
  
  #<stream>
  
  #Format rtp
  
  #File "/usr/local/httpd/htdocs/test1.mpg"
  
  #</stream>
  
  
  
  
  
  ##################################################################
  
  # SDP/multicast examples
  
  #
  
  # If you want to send your stream in multicast, you must set the
  
  # multicast address with MulticastAddress. The port and the TTL can
  
  # also be set.
  
  #
  
  # An SDP file is automatically generated by ffserver by adding the
  
  # &#39;sdp&#39; extension to the stream name (here
  
  # http://localhost:8090/test1-sdp.sdp). You should usually give this
  
  # file to your player to play the stream.
  
  #
  
  # The &#39;NoLoop&#39; option can be used to avoid looping when the stream is
  
  # terminated.
  
  
  
  #<stream>
  
  #Format rtp
  
  #File "/usr/local/httpd/htdocs/test1.mpg"
  
  #MulticastAddress 224.124.0.1
  
  #MulticastPort 5000
  
  #MulticastTTL 16
  
  #NoLoop
  
  #</stream>
  
  
  
  
  
  ##################################################################
  
  # Special streams
  
  
  
  # Server status
  
  
  
  <stream>
  
  Format status
  
  
  
  # Only allow local people to get the status
  
  ACL allow localhost
  
  ACL allow 192.168.0.0 192.168.255.255
  
  
  
  #FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
  
  </stream>
  
  
  
  
  
  # Redirect index.html to the appropriate site
  
  
  
  <redirect>
  
  URL http://www.ffmpeg.org/
  
  </redirect>

  Kindly anyone please assist me whether I am missing something or do i need to change my server.conf file ? I have referred many websites. But still I am unable to fix it. Thanks in advance.