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• ffserver "dimensions not set" when loading stream

  29 mai 2013, par GreenGiant

  I am live streaming a webcam from my raspberry pi using avconv (ffmpeg "replacement")

  avconv -f video4linux2 -v debug -r 5 -s 176x144 -i /dev/video0 -vcodec mjpeg http://192.168.0.3:8090/feed1.ffm

  to my local network OSX machine (for testing) running ffserver

  Port 8090
  
  BindAddress 0.0.0.0
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 10000
  
  CustomLog -
  
  NoDaemon
  
  
  
  <feed>
  
     File feed1.ffm
  
     FileMaxSize 20M
  
     ACL allow 192.168.0.10
  
  </feed>
  
  
  
  <stream>
  
     Feed feed1.ffm
  
     Format mjpeg
  
     NoAudio
  
     VideoQMin 1
  
     VideoQMax 10
  
     VideoSize 176x144
  
     VideoFrameRate 5
  
  </stream>

  When I start avconv it appears to be streaming to ffserver fine :

  Output #0, ffm, to 'http://192.168.0.3:8090/feed1.ffm':
  
    Metadata:
  
      encoder         : Lavf55.0.1
  
      Stream #0.0, 0, 1/1000000: Video: mjpeg, yuvj420p, 320x240, 1/5, q=2-31, 200 kb/s, 1000k tbn, 5 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo -> mjpeg)
  
  Press ctrl-c to stop encoding
  
  frame=  108 fps= 18 q=21.7 size=     688kB time=21.60 bitrate= 260.9kbits/s 

  And the ffserver status page shows the stream

  

  However when I load http://localhost:8090/test.mjpeg in VLC it doesn't play and ffserver spits out :

  Sat May 25 17:25:34 2013 dimensions not set
  
  Sat May 25 17:25:34 2013 Error writing output header
  
  Sat May 25 17:25:34 2013 127.0.0.1 - - [GET] "/test.mjpeg HTTP/1.1" 200 66

  I've tried so many different configurations and settings, I'm at a loss to what is causing that error !

  Thank you

• RTMP to HLS : Segmentation fault

  25 mai 2013, par Wildan Muhlis

  I have tried convert RTMP to HLS with ffmpeg. The segments created without -segment_format mpegts parameter, but it resulting corrupted .m3u8 file, If I add the parameter, segments couldn't created due to below error message.

  Anyone know how to solve it ?

  ffmpeg encoding profile :

  ffmpeg -y -i ${input}" live=1 swfVfy=1" \
  
  -ar 48000 \
  
  -ab 64k \
  
  -s ${WIDTH}x${HEIGHT} \
  
  -vcodec libx264 \
  
  -b:v ${BR} \
  
  -partitions +parti4x4+partp8x8+partb8x8 \
  
  -subq 7 \
  
  -trellis 0 \
  
  -refs 0 \
  
  -coder 0 \
  
  -me_range 16 \
  
  -keyint_min 25 \
  
  -sc_threshold 40 \
  
  -i_qfactor 0.71 \
  
  -bt 200k \
  
  -maxrate ${BR} \
  
  -bufsize ${BR} \
  
  -rc_eq 'blurCplx^(1-qComp)' \
  
  -qcomp 0.6 \
  
  -qmin 30 \
  
  -qmax 51 \
  
  -qdiff 4 \
  
  -level 30 \
  
  -aspect ${WIDTH}:${HEIGHT} \
  
  -g 30 \
  
  -async 2 \
  
  -flags -global_header -map 0 \
  
  -f segment \
  
  -flags +loop \
  
  -segment_time 10 \
  
  -segment_list ${SEGMENT_LIST} \
  
  -segment_format mpegts\
  
  -segment_list_flags +live \
  
  ${OUTPUT}

  Error log :

  ffmpeg version N-53211-g5918b7a Copyright (c) 2000-2013 the FFmpeg developers
  
    built on May 18 2013 09:16:55 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
  
    configuration: --as=yasm --enable-gpl --enable-pthreads --disable-ffserver --disable-shared --enable-static --enable-gpl --enable-libfdk_aac --enable-libmp3lame --enable-libtheora --enable-libvpx --enable-libx264 --enable-librtmp --enable-nonfree
  
    libavutil      52. 33.100 / 52. 33.100
  
    libavcodec     55. 10.100 / 55. 10.100
  
    libavformat    55.  7.100 / 55.  7.100
  
    libavdevice    55.  0.100 / 55.  0.100
  
    libavfilter     3. 68.101 /  3. 68.101
  
    libswscale      2.  3.100 /  2.  3.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  3.100 / 52.  3.100
  
  rtmp server sent error
  
  Metadata:
  
    audiocodecid          .mp3
  
    videokeyframe_frequency5.00
  
    avclevel              30.00
  
    videodevice           ASUS USB2.0 Webcam
  
    audiosamplerate       44100.00
  
    audiochannels         2.00
  
    width                 320.00
  
    videodatarate         200.00
  
    presetname            Custom
  
    audioinputvolume      75.00
  
    creationdate          Sat May 25 20:02:26 2013
  
    videocodecid          avc1
  
    audiodevice           Microphone (Realtek High Defini
  
    avcprofile            66.00
  
    audiodatarate         96.00
  
    height                240.00
  
    framerate             30.00
  
  [flv @ 0x2df2960] max_analyze_duration 5000000 reached at 5018000 microseconds
  
  [flv @ 0x2df2960] decoding for stream 0 failed
  
  Input #0, flv, from 'rtmp://122.22.117.60:1935/oflaDemo/livestream live=1 swfVfy=1':
  
    Metadata:
  
      videokeyframe_frequency: 5
  
      avclevel        : 30
  
      videodevice     : ASUS USB2.0 Webcam
  
      keywords        : 
  
      audiochannels   : 2
  
      presetname      : Custom
  
      copyright       : 
  
      audioinputvolume: 75
  
      creationdate    : Sat May 25 20:02:26 2013
  
                      : 
  
      author          : 
  
      audiodevice     : Microphone (Realtek High Defini
  
      avcprofile      : 66
  
      title           : 
  
      description     : 
  
      rating          : 
  
    Duration: N/A, start: 0.000000, bitrate: 303 kb/s
  
      Stream #0:0: Video: h264 (Baseline), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 204 kb/s, 30 tbr, 1k tbn, 60 tbc
  
      Stream #0:1: Audio: mp3, 44100 Hz, stereo, s16p, 98 kb/s
  
  Segmentation fault

• Ffmpeg send duration of video to client (using node-fluent-ffmpeg)

  26 mai 2013, par Vprnl

  I'm really new to the world of ffmpeg so please excuses me if this is a stupid queston.

  I'm using the module Node-fluent-ffmpeg to stream a movie and convert it from avi to webm with FFMPEG.

  So far so good (it plays the video), but I'm having trouble parsing the duration to the player. It also gives me an error even though I plays the video.

  my code is as followed :

  var stat = fs.statSync(movie);
  
  
  
  var start = 0;
  
  var end = 0;
  
  var range = req.header('Range');
  
  if (range != null) {
  
  start = parseInt(range.slice(range.indexOf('bytes=')+6,
  
    range.indexOf('-')));
  
  end = parseInt(range.slice(range.indexOf('-')+1,
  
    range.length));
  
  }
  
  if (isNaN(end) || end == 0) end = stat.size-1;
  
  if (start > end) return;
  
  
  
  var duration = (end / 1024) * 8 / 1024;
  
  
  
  res.writeHead(206, { // NOTE: a partial http response
  
      'Connection':'close',
  
      'Content-Type':'video/webm',
  
      'Content-Length':end - start,
  
      'Content-Range':'bytes '+start+'-'+end+'/'+stat.size,
  
      'Transfer-Encoding':'chunked'
  
  });
  
  
  
  var proc = new ffmpeg({ source: movie, nolog: true, priority: 1, timeout:15000})
  
      .toFormat('webm')
  
      .addOptions(['-probesize 900000', '-analyzeduration 0', '-minrate 1024k', '-maxrate 1024k', '-bufsize 1835k', '-t '+duration+' -ss'])
  
      .writeToStream(res, function(retcode, error){
  
      if (!error){
  
          console.log('file has been converted succesfully',retcode);
  
      }else{
  
          console.log('file conversion error',error);
  
      }
  
  });

  I set the header with a start and a end based on this article : http://delog.wordpress.com/2011/04/25/stream-webm-file-to-chrome-using-node-js/

  I calculate the length in seconds in the variable duration.

  The error FFmpeg is giving me is :

      file conversion error ffmpeg version N-52458-gaa96439 Copyright (c) 2000-2013 the FFmpeg developers
  
        built on Apr 24 2013 22:19:32 with gcc 4.8.0 (GCC)
  
        configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --e
  
      nable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable
  
      -libgsm --enable-libilbc --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --ena
  
      ble-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwola
  
      me --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enabl
  
      e-libxvid --enable-zlib
  
        libavutil      52. 27.101 / 52. 27.101
  
        libavcodec     55.  6.100 / 55.  6.100
  
        libavformat    55.  3.100 / 55.  3.100
  
        libavdevice    55.  0.100 / 55.  0.100
  
        libavfilter     3. 60.101 /  3. 60.101
  
        libswscale      2.  2.100 /  2.  2.100
  
        libswresample   0. 17.102 /  0. 17.102
  
        libpostproc    52.  3.100 / 52.  3.100
  
      Input #0, avi, from 'C:/temp/test.avi':
  
        Metadata:
  
          encoder         : Nandub v1.0rc2
  
        Duration: 00:01:09.78, start: 0.000000, bitrate: 1517 kb/s
  
          Stream #0:0: Video: msmpeg4v3 (DIV3 / 0x33564944), yuv420p, 640x352, 23.98 tbr, 23.98 tbn, 23.98 tbc
  
          Stream #0:1: Audio: mp3 (U[0][0][0] / 0x0055), 48000 Hz, stereo, s16p, 222 kb/s
  
      [libvpx @ 0036db20] v1.2.0
  
      Output #0, webm, to 'pipe:1':
  
        Metadata:
  
          encoder         : Lavf55.3.100
  
          Stream #0:0: Video: vp8, yuv420p, 640x352, q=-1--1, 200 kb/s, 1k tbn, 23.98 tbc
  
          Stream #0:1: Audio: vorbis, 48000 Hz, stereo, fltp
  
      Stream mapping:
  
        Stream #0:0 -> #0:0 (msmpeg4 -> libvpx)
  
        Stream #0:1 -> #0:1 (mp3 -> libvorbis)

  The client side player (which is VideoJs) says the file is infinite/NaN in length.

  I feel like I'm pretty close to a solution but my inexperience with the subject matter prohibits me from getting it to work. If I'm unclear in any way please let me know. (I have a tendency of explaining things fuzzy.)

  Thanks in advance !

  [EDIT]

  I removed the duration bit because it has nothing to do with the issue. I checked the response header of the client and saw :

   Accept-Ranges:bytes
  
   Connection:keep-alive
  
   Content-Length:13232127
  
   Content-Range:bytes 0-13232127/13232128
  
   Content-Type:video/webm

  Why can't the client figure out the duration even though it receives it in the header ?