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• Evolution #2291 : Faire une redirection 301 pour les anciennes url

  6 septembre 2011, par Bruno CAILLARD

  Pas avec les url aborescentes : Exemple : Anciennes url : http://www.terrasse-bois.info/bois-de-terrasse ou http://www.terrasse-bois.info/bois-de-terrasse.html devraient être redirigées vers Nouvelle url : http://www.terrasse-bois.info/choisir-les-materiaux-pour-votre/bois-de-terrasse Cela (...)

• flv reencode to mp4 for iphone/ipod via ffmpeg and x264 (quality issue)

  3 octobre 2011, par zeroasterisk

  There are a lot of questions on this topic, and I've read most of them and most of the google search results I could come up with.

  When I use FFMPEG to convert a FLV to a iphone3 compatble MP4 file, it just doesn't preserver enough of the quality. Yes, I've worked the hell out of -sameq and -b and -bt settings, text just isn't readable.

  Next I tried to split the video out and process it directly, using these instructions :
  https://sites.google.com/site/linuxencoding/x264-encoding-guide

  The problem is myplayer (via ffmpeg) was not able to determine the duration of the FLV (even though the metadata was set).

  (I assume) Because of that unknown duration, when I create the MP4 file, the resulting x264 file plays through super-fast while the audio plays at the normal rate.

  user@server:/tmp# mplayer -nosound -benchmark -sws 9 -vf dsize=640:480:0,scale=0:0,expand=640:480 -vo yuv4mpeg:file=>(x264 --demuxer y4m --crf 0 --preset slow --threads auto --output output.264 - 2>x264.log) 'input.flv'
  
  MPlayer 1.0rc4-4.4.5 (C) 2000-2010 MPlayer Team
  
  mplayer: could not connect to socket
  
  mplayer: No such file or directory
  
  Failed to open LIRC support. You will not be able to use your remote control.
  
  
  
  Playing input.flv.
  
  libavformat file format detected.
  
  [flv @ 0x1202460]Estimating duration from bitrate, this may be inaccurate
  
  [lavf] stream 0: video (vp6f), -vid 0
  
  [lavf] stream 1: audio (nellymoser), -aid 0
  
  VIDEO:  [VP6F]  1680x992  0bpp  1000.000 fps   33.4 kbps ( 4.1 kbyte/s)
  
  Clip info:
  
   audiocodecid: 6
  
   audiodatarate: 86
  
   audiosamplerate: 44100
  
   audiosamplesize: 16
  
   audiosize: 6097005
  
   canSeekToEnd: true
  
   datasize: 8609138
  
   duration: 567
  
   framerate: 2
  
   hasAudio: true
  
   hasCuePoints: false
  
   hasKeyframes: true
  
   hasMetadata: true
  
   hasVideo: true
  
   height: 992
  
   lasttimestamp: 567
  
   metadatacreator: flvtool++ (Facebook, Motion project, dweatherford)
  
   stereo: false
  
   totalframes: 1043
  
   videocodecid: 4
  
   videodatarate: 33
  
   videosize: 2316256
  
   width: 1680
  
  Using (default) progressive frame mode.Opening video filter: [expand w=640 h=480]
  
  Expand: 640 x 480, -1 ; -1, osd: 0, aspect: 0.000000, round: 1
  
  Opening video filter: [scale w=0 h=0]
  
  Opening video filter: [dsize=640:480:0]
  
  ==========================================================================
  
  Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
  
  Selected video codec: [ffvp6f] vfm: ffmpeg (FFmpeg VP6 Flash)
  
  ==========================================================================
  
  Audio: no sound
  
  Starting playback...
  
  Movie-Aspect is undefined - no prescaling applied.
  
  [swscaler @ 0x7f0c738b9620]Lanczos scaler, from yuv420p to yuv420p using MMX2
  
  VO: [yuv4mpeg] 640x480 => 641x480 Planar YV12

  I have also tried specifying FPS, but no change in results

  user@server:/tmp# mplayer -nosound -fps 25-benchmark -sws 9 -vf dsize=640:480:0,scale=0:0,expand=640:480 -vo yuv4mpeg:file=>(x264 --demuxer y4m --fps 25 --crf 0 --preset slow --threads auto --output output.264 - 2>x264.log) 'input.flv'

  Can someone tell me how to either :

  1. fix my split A/V processing/timing/duration issues ?
  2. improve the
     quality of the FFMPEG conversion of FLV to iphone3 compatible
     format ?

• Streaming without Content-Length in response

  29 août 2011, par kain

  I'm using Node.js, Express (and connect), and fluent-ffmpeg.

  We want to stream audio files that are stored on Amazon S3 through http.

  We have all working, except that we would like to add a feature, the on-the-fly conversion of the stream through ffmpeg.

  This is working well, the problem is that some browsers checks in advance before actually getting the file.

  Incoming requests containing the Range header, for which we reply with a 206 with all the info from S3, have a fundamental problem : we need to know in advance the content-length of the file.

  We don't know that since it is going through ffmpeg.

  One solution might be to write out the resulting content-length directly on S3 when storing the file (in a special header), but this means we have to go through the pain of having queues to encode after upload just to know the size for future requests.
  It also means that if we change compressor or preset we have to go through all this over again, so it is not a viable solution.

  We also noticed big differencies in the way Chrome and Safari request the audio tag src, but this may be discussion for another topic.

  Fact is that without a proper content-length header in response everything seems to break or browsers goes in an infinite loop or restart the stream at pleasure.

  Ideas ?