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• PHP - Problems running ffmpeg.exe with shell_exec on Windows

  30 septembre 2012, par King

  I am trying to encode an FLV file to MP4 with ffmpeg.exe, its all working fine when running the code on the commandprompt (I have tried simply echoing the command and pasting it into the cmdline)

  Note : Using windows.

  Here is the code :
  it does not return anything, and does not encode anything, there are no errors, nothing happens :(, system($cmd, $returnval) returns 1 that's all
  running commands like dir works fine, so I have the required access.

  <?php
  
  $title = $_GET['file'];
  
  // encoding command > , below shows info "flv/'.$title.'.flv" "mp4/'.$title.'.mp4"
  
  $cmd = ('cd C:/inetpub/wwwroot/run/ && ffmpeg.exe -i "flv/'.$title.'.flv"');
  
  $ll = system($cmd, $fv);
  
  echo "CMD: $cmd \nFull output: $fv \nLL = $ll";
  
  ?>

  ffmpeg.exe -h will return the help text, I am starting to wonder if it could have something to do with the other outputs having colours in them ?

  Any ideas ? :O

  (The cd is not the problem because $cmd = ('cd C:/inetpub/wwwroot/run/ && ffmpeg.exe -h'); works fine, its actually already in that directory)

• Something wrong with my m3u8 bandwidth value

  6 octobre 2012, par Jason

  I use ffmpeg to encode my sample videos following the recommanded bitrates in Technical Note TN2224, then use HLS tools to segment it and create playlists, finally create the variant plist file "all.m3u8"
  I used the validation tool to validate my HLS content, it ended up showing except for the 64k audio only bandwidth is low, others are stay in the same bandwidth, I opened "all.m3u8" using text editor and seeing that all other bitrate contents are using the same bandwidth. No matter how I change parameters in the ffmpeg command, I still can't correct them. The following command is the one I used to encode contents :
  ffmpeg -i input.m4v -acodec libfaac -vcodec libx264 -s 480x360 -b 350k -r 29.97 -vpre medium output.mp4

  The following command is for generating the segments and plists :
  mediafilesegmenter -b http://www.example.com/stream/ -I -f ~/Documents/sample/ output.mp4

  The following command is for generating the all.m3u8 :
  

  variantplaylistcreator -o all.m3u8 http://www.example.com/stream/110/prog_index.m3u8 ~/Documents/sample/110/prog_index.m3u8 -iframe-url http://www.freeyourteam.com/stream/110/iframe_index.m3u8 http://www.example.com/stream/200/prog_index.m3u8 ~/Documents/sample/200/prog_index.m3u8 -iframe-url http://www.freeyourteam.com/stream/200/iframe_index.m3u8 http://www.example.com/stream/350/prog_index.m3u8 ~/Documents/sample/350/prog_index.m3u8 -iframe-url http://www.freeyourteam.com/stream/350/iframe_index.m3u8 http://www.example.com/stream/550/prog_index.m3u8 ~/Documents/sample/550/prog_index.m3u8 -iframe-url http://www.freeyourteam.com/stream/550/iframe_index.m3u8 http://www.example.com/stream/64/prog_index.m3u8 ~/Documents/sample/64/prog_index.m3u8

  and in my "all.m3u8", the bandwidths are all 523894 :
  

  Please allow me to ask two more basic questions :
  In the tech note, recommanded bitrates are 64 Kbps, 110 Kbps, 200 Kbps, 350 Kbps, 550 Kbps, I wonder if this value includes the audio bitrate or exclude the audio.
  How do you insert keyframe to segment ? Because in the document it says :"You must include at least one keyframe per segment, preferably more. If you only include one, put it at the beginning of the segment." I don't quite get how you can do it.
  Thank you very much for your help and I do appreciate your time.

• gstreamer records

  13 octobre 2012, par Eric

  I use the following command to record audio and video from my webcam

  gst-launch-0.10 v4l2src ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! \
  
               tee name=t_vid ! queue ! videoflip method=horizontal-flip ! \
  
               xvimagesink sync=false t_vid. ! queue ! \
  
               videorate ! video/x-raw-yuv,framerate=30/1 ! queue ! mux. \
  
               autoaudiosrc ! audiorate ! audio/x-raw-int,rate=48000,channels=1,depth=16 ! queue ! \
  
               audioconvert ! queue ! mux. avimux name=mux ! \
  
               filesink location=video.avi

  And the result is correct in term of synchronicity between the flows. However the avi file is very big since that's uncompressed data...
  Could you advice me howto reduce the size of the records. Note that I after recording I split audio and video in separate files for processing. It is crucial to keep the synchronicity.

  * Edit *

  I tried to use ffmpeg to compress the avi files using this command :

  ffmpeg -i video.avi -vcodec msmpeg4v2 output.avi

  But it seems that bitrate is invalid (N/A since its raw data ?)
  Here is the output :

  Input #0, avi, from 'video.avi':
  
  Duration: 00:00:00.00, start: 0.000000, bitrate: -2147483 kb/s
  
    Stream #0.0: Video: rawvideo, yuv420p, 640x480, 30 tbr, 30 tbn, 30 tbc
  
    Stream #0.1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s
  
  [buffer @ 0xef57e0] w:640 h:480 pixfmt:yuv420p
  
  Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt'
  
  [ac3 @ 0xedece0] channel_layout not specified
  
  [ac3 @ 0xedece0] No channel layout specified. The encoder will guess the layout, but it     might be incorrect.
  
  [ac3 @ 0xedece0] invalid bit rate
  
  Output #0, avi, to 'output.avi':
  
    Stream #0.0: Video: msmpeg4v2, yuv420p, 640x480, q=2-31, 200 kb/s, 90k tbn, 30 tbc
  
    Stream #0.1: Audio: ac3, 48000 Hz, mono, flt, 200 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
    Stream #0.1 -> #0.1
  
  Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height

  Thanks for helping.