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• ffmpeg ffserver - create a mosaic from two 720p webcam feeds

  28 juillet 2015, par der_felix

  for a project i would like to take the video feeds (NO audio) of two logitech c920 webcams, put them side-by-side and stream them.
  the c920 is able to compress the video feed with h264 itself(if enabled) and delivers 1080p with upto 30fps.
  the stream is then loaded in an android app by a ffmpeg library and rendered to the screen.

  what i already know :
  i know that i can take multiple streams or input files and create a mosaic stream via the filter_complex module.
  http and h264 seem to be good for streaming, but other configurations are also welcom if they are faster/better.

  the question :
  but how can i start the cameras with v4l2, set the camera resoltution and camera internal encoding and use these streams to create the mosaic ?
  the mosaic should be unscaled (=2560x720px).

  and i very often get the error code 256 but didnt find a solution what it means.

  the system : laptop with usb3, ubuntu 15.04x64 ffmpeg 2.7.1 and ffserver 2.5.7

  thanks for your help

  ffserver config :

  HTTPPort 8080                
  
  HTTPBindAddress 0.0.0.0      
  
  MaxHTTPConnections 2000   
  
  MaxClients 1000         
  
  MaxBandwidth 50000
  
  CustomLog -      
  
  #NoDaemon       
  
  
  
  <feed>         
  
  
  
  File /tmp/feed1.ffm
  
  Launch ffmpeg -f v4l2 - input_format h264 -i /dev/video0 -i /dev/video1 -size 1280x720 -r 30 -filter_complex "nullsrc=size=2560x720 [base]; [0:v] setpts=PTS-STARTPTS [left]; [1:v] setpts=PTS-STARTPTS [right]; [base][left] overlay=shortest=1 [tmp1]; [tmp1][right] overlay=shortest=1:x=1280"  -c:v libx264 -f mpegts
  
  
  
  </feed>
  
  
  
  <stream>
  
  
  
  Feed feed1.ffm
  
  Format mpegts      
  
  VideoBitRate 1024   
  
  #VideoBufferSize 1024
  
  VideoFrameRate 30      
  
  #VideoSize hd720      
  
  VideoSize 2560x720
  
  #VideoIntraOnly         
  
  #VideoGopSize 12      
  
  VideoCodec libx264      
  
  NoAudio            
  
  VideoQMin 3         
  
  VideoQMax 31
  
  NoDefaults
  
  
  
  </stream>
  
  
  
  <stream>        
  
     Format status
  
     #Only allow local people to get the status
  
     ACL allow localhost
  
     ACL allow 192.168.0.0 192.168.255.255
  
  </stream>

  output :

  ubuntu@ubuntu:~$ ffserver
  
  ffserver version 2.5.7-0ubuntu0.15.04.1 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
  
    libavutil      54. 15.100 / 54. 15.100
  
    libavcodec     56. 13.100 / 56. 13.100
  
    libavformat    56. 15.102 / 56. 15.102
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  2.103 /  5.  2.103
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Tue Jul 28 10:13:44 2015 FFserver started.
  
  Tue Jul 28 10:13:44 2015 Launch command line: ffmpeg -f v4l2 - input_format h264 -i /dev/video0 -i /dev/video1 -size 1280x720 -r 30 -filter_complex nullsrc=size=2560x720 [base]; [0:v] setpts=PTS-STARTPTS [left]; [1:v] setpts=PTS-STARTPTS [right]; [base][left] overlay=shortest=1 [tmp1]; [tmp1][right] overlay=shortest=1:x=1280 -c:v libx264 -f mpegts http://127.0.0.1:8080/feed1.ffm
  
  feed1.ffm: Pid 17388 exited with status 256 after 0 seconds

  Hey guys !

  Here is our plan b for the mosaic stream.

  Alternative config :

  HTTPPort 8080         
  
  HTTPBindAddress 0.0.0.0   
  
  MaxHTTPConnections 2000   
  
  MaxClients 1000         
  
  MaxBandwidth 50000      
  
  CustomLog -         
  
  
  
  <feed>
  
  File /tmp/feedlinks.ffm
  
  Launch ffmpeg -f v4l2 -input_format h264 -vcodec h264 -i /dev/video0 -video_size 1280x720 -r 30
  
  </feed>
  
  
  
  <feed>
  
  File /tmp/feedrechts.ffm
  
  Launch ffmpeg -f v4l2 -input_format h264 -vcodec h264 -i /dev/video1 -video_size 1280x720 -r 30 
  
  </feed>
  
  
  
  <stream>
  
  Feed feedlinks.ffm
  
  Format mpegts
  
  VideoBitRate 512
  
  VideoFrameRate 30
  
  VideoSize hd720
  
  VideoCodec libx264
  
  NoAudio
  
  VideoQMin 3
  
  VideoQMax 31
  
  </stream>
  
  
  
  <stream>
  
  Feed feedrechts.ffm
  
  Format mpegts
  
  VideoBitRate 512
  
  VideoFrameRate 30
  
  VideoSize hd720
  
  VideoCodec libx264
  
  NoAudio
  
  VideoQMin 3
  
  VideoQMax 31
  
  </stream>
  
  
  
  <feed>
  
  File /tmp/feedmosaic.ffm
  
  Launch ffmpeg -i http://localhost:8080/testlinks.mpg -i http://localhost:8080/testrechts.mpg -filter_complex "nullsrc=size=2560x720 [base]; [0:v] setpts=PTS-STARTPTS [left]; [1:v] setpts=PTS-STARTPTS [right]; [base][left] overlay=shortest=1 [tmp1]; [tmp1][right] overlay=shortest=1:x=1280" -c:v libx264 -preset ultrafast -f mpegts
  
  </feed>
  
  
  
  <stream>
  
  Feed feedmosaic.ffm
  
  Format mpegts         # Format of the stream
  
  VideoFrameRate 30      # Number of frames per second
  
  VideoSize 2560x720
  
  VideoCodec libx264      # Choose your codecs.
  
  NoAudio            # Suppress audio
  
  VideoQMin 3         # Videoquality ranges from 1 - 31 (worst to best)
  
  VideoQMax 31
  
  NoDefaults
  
  </stream>
  
  
  
  <stream>           # Server status URL
  
     Format status
  
     # Only allow local people to get the status
  
     ACL allow localhost
  
     ACL allow 192.168.0.0 192.168.255.255
  
     ACL allow 192.168.178.0 192.168.255.255
  
  </stream>

  And this is the new output :

  ubuntu@ubuntu:~$ ffserver
  
  ffserver version 2.5.7-0ubuntu0.15.04.1 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
  
    libavutil      54. 15.100 / 54. 15.100
  
    libavcodec     56. 13.100 / 56. 13.100
  
    libavformat    56. 15.102 / 56. 15.102
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  2.103 /  5.  2.103
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  /etc/ffserver.conf:44: Setting default value for video bit rate tolerance = 128000. Use NoDefaults to disable it.
  
  /etc/ffserver.conf:44: Setting default value for video rate control equation = tex^qComp. Use NoDefaults to disable it.
  
  /etc/ffserver.conf:44: Setting default value for video max rate = 1024000. Use NoDefaults to disable it.
  
  /etc/ffserver.conf:44: Setting default value for video buffer size = 1024000. Use NoDefaults to disable it.
  
  /etc/ffserver.conf:61: Setting default value for video bit rate tolerance = 128000. Use NoDefaults to disable it.
  
  /etc/ffserver.conf:61: Setting default value for video rate control equation = tex^qComp. Use NoDefaults to disable it.
  
  /etc/ffserver.conf:61: Setting default value for video max rate = 1024000. Use NoDefaults to disable it.
  
  /etc/ffserver.conf:61: Setting default value for video buffer size = 1024000. Use NoDefaults to disable it.
  
  Tue Jul 28 11:13:01 2015 Codec bitrates do not match for stream 0
  
  Tue Jul 28 11:13:01 2015 FFserver started.
  
  Tue Jul 28 11:13:01 2015 Launch command line: ffmpeg -f v4l2 -input_format h264 -vcodec h264 -i /dev/video0 -video_size 1280x720 -r 30 http://127.0.0.1:8080/feedlinks.ffm
  
  Tue Jul 28 11:13:01 2015 Launch command line: ffmpeg -f v4l2 -input_format h264 -vcodec h264 -i /dev/video1 -video_size 1280x720 -r 30 http://127.0.0.1:8080/feedrechts.ffm
  
  Tue Jul 28 11:13:01 2015 Launch command line: ffmpeg -i http://localhost:8080/testlinks.mpg -i http://localhost:8080/testrechts.mpg -filter_complex nullsrc=size=2560x720 [base]; [0:v] setpts=PTS-STARTPTS [left]; [1:v] setpts=PTS-STARTPTS [right]; [base][left] overlay=shortest=1 [tmp1]; [tmp1][right] overlay=shortest=1:x=1280 -c:v libx264 -preset ultrafast -f mpegts http://127.0.0.1:8080/feedmosaic.ffm
  
  Tue Jul 28 11:13:02 2015 127.0.0.1 - - [GET] "/feedlinks.ffm HTTP/1.1" 200 4175
  
  Tue Jul 28 11:13:02 2015 127.0.0.1 - - [GET] "/feedrechts.ffm HTTP/1.1" 200 4175
  
  Tue Jul 28 11:13:18 2015 127.0.0.1 - - [POST] "/feedmosaic.ffm HTTP/1.1" 200 4096
  
  Tue Jul 28 11:13:18 2015 127.0.0.1 - - [GET] "/testlinks.mpg HTTP/1.1" 200 2130291
  
  Tue Jul 28 11:13:18 2015 127.0.0.1 - - [GET] "/testrechts.mpg HTTP/1.1" 200 1244999
  
  feedmosaic.ffm: Pid 18775 exited with status 256 after 17 seconds

  Thanks for your help !

• Compute PTS and DTS correctly to sync audio and video ffmpeg C++

  14 août 2015, par Kaidul Islam

  I am trying to mux H264 encoded data and G711 PCM data into mov multimedia container. I am creating AVPacket from encoded data and initially the PTS and DTS value of video/audio frames is equivalent to AV_NOPTS_VALUE. So I calculated the DTS using current time information. My code -

  bool AudioVideoRecorder::WriteVideo(const unsigned char *pData, size_t iDataSize, bool const bIFrame) {
  
      .....................................
  
      .....................................
  
      .....................................
  
      AVPacket pkt = {0};
  
      av_init_packet(&pkt);
  
      int64_t dts = av_gettime();
  
      dts = av_rescale_q(dts, (AVRational){1, 1000000}, m_pVideoStream->time_base);
  
      int duration = 90000 / VIDEO_FRAME_RATE;
  
      if(m_prevVideoDts > 0LL) {
  
          duration = dts - m_prevVideoDts;
  
      }
  
      m_prevVideoDts = dts;
  
  
  
      pkt.pts = AV_NOPTS_VALUE;
  
      pkt.dts = m_currVideoDts;
  
      m_currVideoDts += duration;
  
      pkt.duration = duration;
  
      if(bIFrame) {
  
          pkt.flags |= AV_PKT_FLAG_KEY;
  
      }
  
      pkt.stream_index = m_pVideoStream->index;
  
      pkt.data = (uint8_t*) pData;
  
      pkt.size = iDataSize;
  
  
  
      int ret = av_interleaved_write_frame(m_pFormatCtx, &pkt);
  
  
  
      if(ret < 0) {
  
          LogErr("Writing video frame failed.");
  
          return false;
  
      }
  
  
  
      Log("Writing video frame done.");
  
  
  
      av_free_packet(&pkt);
  
      return true;
  
  }
  
  
  
  bool AudioVideoRecorder::WriteAudio(const unsigned char *pEncodedData, size_t iDataSize) {
  
      .................................
  
      .................................
  
      .................................
  
      AVPacket pkt = {0};
  
      av_init_packet(&pkt);
  
  
  
      int64_t dts = av_gettime();
  
      dts = av_rescale_q(dts, (AVRational){1, 1000000}, (AVRational){1, 90000});
  
      int duration = AUDIO_STREAM_DURATION; // 20
  
      if(m_prevAudioDts > 0LL) {
  
          duration = dts - m_prevAudioDts;
  
      }
  
      m_prevAudioDts = dts;
  
      pkt.pts = AV_NOPTS_VALUE;
  
      pkt.dts = m_currAudioDts;
  
      m_currAudioDts += duration;
  
      pkt.duration = duration;
  
  
  
      pkt.stream_index = m_pAudioStream->index;
  
      pkt.flags |= AV_PKT_FLAG_KEY;
  
      pkt.data = (uint8_t*) pEncodedData;
  
      pkt.size = iDataSize;
  
  
  
      int ret = av_interleaved_write_frame(m_pFormatCtx, &pkt);
  
      if(ret < 0) {
  
          LogErr("Writing audio frame failed: %d", ret);
  
          return false;
  
      }
  
  
  
      Log("Writing audio frame done.");
  
  
  
      av_free_packet(&pkt);
  
      return true;
  
  }

  And I added stream like this -

  AVStream* AudioVideoRecorder::AddMediaStream(enum AVCodecID codecID) {
  
      ................................
  
      .................................   
  
      pStream = avformat_new_stream(m_pFormatCtx, codec);
  
      if (!pStream) {
  
          LogErr("Could not allocate stream.");
  
          return NULL;
  
      }
  
      pStream->id = m_pFormatCtx->nb_streams - 1;
  
      pCodecCtx = pStream->codec;
  
      pCodecCtx->codec_id = codecID;
  
  
  
      switch(codec->type) {
  
      case AVMEDIA_TYPE_VIDEO:
  
          pCodecCtx->bit_rate = VIDEO_BIT_RATE;
  
          pCodecCtx->width = PICTURE_WIDTH;
  
          pCodecCtx->height = PICTURE_HEIGHT;
  
          pStream->time_base = (AVRational){1, 90000};
  
          pStream->avg_frame_rate = (AVRational){90000, 1};
  
          pStream->r_frame_rate = (AVRational){90000, 1}; // though the frame rate is variable and around 15 fps
  
          pCodecCtx->pix_fmt = STREAM_PIX_FMT;
  
          m_pVideoStream = pStream;
  
          break;
  
  
  
      case AVMEDIA_TYPE_AUDIO:
  
          pCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
  
          pCodecCtx->bit_rate = AUDIO_BIT_RATE;
  
          pCodecCtx->sample_rate = AUDIO_SAMPLE_RATE;
  
          pCodecCtx->channels = 1;
  
          m_pAudioStream = pStream;
  
          break;
  
  
  
      default:
  
          break;
  
      }
  
  
  
      /* Some formats want stream headers to be separate. */
  
      if (m_pOutputFmt->flags & AVFMT_GLOBALHEADER)
  
          m_pFormatCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  
  
      return pStream;
  
  }

  There are several problems with this calculation :

  1. The video is laggy and lags behind than audio increasingly with time.

  2. Suppose, an audio frame is received (WriteAudio(..)) little lately like 3 seconds, then the late frame should be started playing with 3 second delay, but it’s not. The delayed frame is played consecutively with previous frame.

  3. Sometimes I recorded for 40 seconds but the file duration is much like 2 minutes, but audio/video is played only few moments like 40 seconds and rest of the file contains nothing and seekbar jumps at en immediately after 40 seconds (tested in VLC).

  EDIT :

  According to Ronald S. Bultje’s suggestion, what I’ve understand :

  m_pAudioStream->time_base = (AVRational){1, 9000}; // actually no need to set as 9000 is already default value for audio as you said
  
  m_pVideoStream->time_base = (AVRational){1, 9000};

  should be set as now both audio and video streams are now in same time base units.

  And for video :

  ...................
  
  ...................
  
  
  
  int64_t dts = av_gettime(); // get current time in microseconds
  
  dts *= 9000; 
  
  dts /= 1000000; // 1 second = 10^6 microseconds
  
  pkt.pts = AV_NOPTS_VALUE; // is it okay?
  
  pkt.dts = dts;
  
  // and no need to set pkt.duration, right?

  And for audio : (exactly same as video, right ?)

  ...................
  
  ...................
  
  
  
  int64_t dts = av_gettime(); // get current time in microseconds
  
  dts *= 9000; 
  
  dts /= 1000000; // 1 second = 10^6 microseconds
  
  pkt.pts = AV_NOPTS_VALUE; // is it okay?
  
  pkt.dts = dts;
  
  // and no need to set pkt.duration, right?

  And I think they are now like sharing same currDts, right ? Please correct me if I am wrong anywhere or missing anything.

  Also, if I want to use video stream time base as (AVRational){1, frameRate} and audio stream time base as (AVRational){1, sampleRate}, how the correct code should look like ?

  EDIT 2.0 :

  m_pAudioStream->time_base = (AVRational){1, VIDEO_FRAME_RATE};
  
  m_pVideoStream->time_base = (AVRational){1, VIDEO_FRAME_RATE};

  And

  bool AudioVideoRecorder::WriteAudio(const unsigned char *pEncodedData, size_t iDataSize) {
  
      ...........................
  
      ......................
  
      AVPacket pkt = {0};
  
      av_init_packet(&pkt);
  
  
  
      int64_t dts = av_gettime() / 1000; // convert into millisecond
  
      dts = dts * VIDEO_FRAME_RATE;
  
      if(m_dtsOffset < 0) {
  
          m_dtsOffset = dts;
  
      }
  
  
  
      pkt.pts = AV_NOPTS_VALUE;
  
      pkt.dts = (dts - m_dtsOffset);
  
  
  
      pkt.stream_index = m_pAudioStream->index;
  
      pkt.flags |= AV_PKT_FLAG_KEY;
  
      pkt.data = (uint8_t*) pEncodedData;
  
      pkt.size = iDataSize;
  
  
  
      int ret = av_interleaved_write_frame(m_pFormatCtx, &pkt);
  
      if(ret < 0) {
  
          LogErr("Writing audio frame failed: %d", ret);
  
          return false;
  
      }
  
  
  
      Log("Writing audio frame done.");
  
  
  
      av_free_packet(&pkt);
  
      return true;
  
  }
  
  
  
  bool AudioVideoRecorder::WriteVideo(const unsigned char *pData, size_t iDataSize, bool const bIFrame) {
  
      ........................................
  
      .................................
  
      AVPacket pkt = {0};
  
      av_init_packet(&pkt);
  
      int64_t dts = av_gettime() / 1000;
  
      dts = dts * VIDEO_FRAME_RATE;
  
      if(m_dtsOffset < 0) {
  
          m_dtsOffset = dts;
  
      }
  
      pkt.pts = AV_NOPTS_VALUE;
  
      pkt.dts = (dts - m_dtsOffset);
  
  
  
      if(bIFrame) {
  
          pkt.flags |= AV_PKT_FLAG_KEY;
  
      }
  
      pkt.stream_index = m_pVideoStream->index;
  
      pkt.data = (uint8_t*) pData;
  
      pkt.size = iDataSize;
  
  
  
      int ret = av_interleaved_write_frame(m_pFormatCtx, &pkt);
  
  
  
      if(ret < 0) {
  
          LogErr("Writing video frame failed.");
  
          return false;
  
      }
  
  
  
      Log("Writing video frame done.");
  
  
  
      av_free_packet(&pkt);
  
      return true;
  
  }

  Is the last change okay ? The video and audio seems synced. Only problem is - the audio is played without the delay regardless the packet arrived in delay.
  Like -

  packet arrival : 1 2 3 4... (then next frame arrived after 3 sec) .. 5

  audio played : 1 2 3 4 (no delay) 5

  EDIT 3.0 :

  zeroed audio sample data :

  AVFrame* pSilentData;
  
  pSilentData = av_frame_alloc();
  
  memset(&pSilentData->data[0], 0, iDataSize);
  
  
  
  pkt.data = (uint8_t*) pSilentData;
  
  pkt.size = iDataSize;
  
  
  
  av_freep(&pSilentData->data[0]);
  
  av_frame_free(&pSilentData);

  Is this okay ? But after writing this into file container, there are dot dot noise during playing the media. Whats the problem ?

  EDIT 4.0 :

  Well, For µ-Law audio the zero value is represented as 0xff. So -

  memset(&pSilentData->data[0], 0xff, iDataSize);

  solve my problem.

• ffmpeg + ffserver : "Broken ffmpeg default settings detected"

  18 octobre 2012, par Chris Nolet

  I'm just trying to connect ffmpeg to ffserver and stream rawvideo.

  I keep getting the error : broken ffmpeg default settings detected from libx264 and then Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height from ffmpeg before it exits.

  I'm launching ffmpeg with the command : ffmpeg -f x11grab -s 320x480 -r 10 -i :0.0 -tune zerolatency http://localhost:8090/feed1.ffm

  My ffserver.conf file (for ffserver) looks like this :

  Port 8090
  
  BindAddress 0.0.0.0
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 1000
  
  CustomLog -
  
  NoDaemon
  
  
  
  <feed>
  
    ACL allow 127.0.0.1
  
  </feed>
  
  
  
  <stream>
  
    Feed feed1.ffm
  
    Format asf
  
  
  
    NoAudio
  
  
  
    VideoBitRate 128
  
    VideoBufferSize 400
  
    VideoFrameRate 24
  
    VideoSize 320x480
  
  
  
    VideoGopSize 12
  
  
  
    VideoQMin 1
  
    VideoQMax 31
  
  
  
    VideoCodec libx264
  
  </stream>
  
  
  
  <stream>
  
    Format status
  
  </stream>

  And the full output is :

  ffmpeg version N-45614-g364c60b Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Oct 17 2012 04:34:04 with Apple clang version 4.1 (tags/Apple/clang-421.11.65) (based on LLVM 3.1svn)
  
    configuration: --enable-shared --enable-libx264 --enable-libmp3lame --enable-x11grab --enable-gpl --enable-version3 --enable-nonfree --enable-hardcoded-tables --cc=/usr/bin/clang --host-cflags='-Os -w -pipe -march=native -Qunused-arguments -mmacosx-version-min=10.7' --extra-cflags='-x objective-c' --extra-ldflags='-framework Foundation -framework Cocoa -framework CoreServices -framework ApplicationServices -lobjc'
  
    libavutil      51. 76.100 / 51. 76.100
  
    libavcodec     54. 66.100 / 54. 66.100
  
    libavformat    54. 32.101 / 54. 32.101
  
    libavdevice    54.  3.100 / 54.  3.100
  
    libavfilter     3. 19.103 /  3. 19.103
  
    libswscale      2.  1.101 /  2.  1.101
  
    libswresample   0. 16.100 /  0. 16.100
  
    libpostproc    52.  1.100 / 52.  1.100
  
  [x11grab @ 0x7f87dc01e200] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 320 height: 480
  
  [x11grab @ 0x7f87dc01e200] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, x11grab, from ':0.0':
  
    Duration: N/A, start: 1350517708.386699, bitrate: 49152 kb/s
  
      Stream #0:0: Video: rawvideo (BGRA / 0x41524742), bgra, 320x480, 49152 kb/s, 10 tbr, 1000k tbn, 10 tbc
  
  [tcp @ 0x7f87dc804120] TCP connection to localhost:8090 failed: Connection refused
  
  [tcp @ 0x7f87dc804b20] TCP connection to localhost:8090 failed: Connection refused
  
  [libx264 @ 0x7f87dd801000] broken ffmpeg default settings detected
  
  [libx264 @ 0x7f87dd801000] use an encoding preset (e.g. -vpre medium)
  
  [libx264 @ 0x7f87dd801000] preset usage: -vpre <speed> -vpre <profile>
  
  [libx264 @ 0x7f87dd801000] speed presets are listed in x264 --help
  
  [libx264 @ 0x7f87dd801000] profile is optional; x264 defaults to high
  
  Output #0, ffm, to &#39;http://localhost:8090/feed1.ffm&#39;:
  
    Metadata:
  
      creation_time   : now
  
      Stream #0:0: Video: h264, yuv420p, 160x128, q=2-31, 128 kb/s, 1000k tbn, 10 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo -> libx264)
  
  Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
  
  </profile></speed>

  Any help much appreciated :)