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• Kdialog unable to cancel job with ffmpeg from progressbar , with other command it works :S

  2 janvier 2015, par user1088530

  i did a script to convert my stuff with ffmpeg

  it does :

  1 create progressbar with cancel button
  2 loop to convert files
  3 set the progress value

  it works fine but it doens’t stop ffmpeg when i click on cancel

  the strange thing is that when i use another program like echo it works like it should it seems a ffmpeg issue

  can onyone help me how to find out the trick ?

  listoffile="$HOME/ffmpeglist.lst"
  
  numberoffile=$(wc -l <$listoffile)
  
  ffmpegpath="/usr/bin/ffmpeg"
  
  
  
      mystufvariabletouseonmyscript=$(kdialog --progressbar "hello this is a progress bar with 100 steps" 100) ;  sleep 2 ; qdbus $mystufvariabletouseonmyscript  showCancelButton true
  
  
  
  
  
  
  
  
  
      until test "true" = `qdbus  $mystufvariabletouseonmyscript wasCancelled` ; do 
  
  
  
              while read line ; do
  
                  qdbus $mystufvariabletouseonmyscript org.kde.kdialog.ProgressDialog.setLabelText $"Starting Conversion...processing file $line"
  
  
  
  
  
                  $ffmpegpath -i   "$line"   "${line%%.*}.$mp3" 
  
  
  
  
  
                  value=$(( $(($a*100))/$numberoffile ))
  
                  qdbus  $mystufvariabletouseonmyscript Set org.kde.kdialog.ProgressDialog value $value
  
                  a=$(($a+1))
  
  
  
                  done< "$listoffile"
  
  
  
          done

• Transcoding Mp4 video on Android with FFMPEG

  18 octobre 2012, par geeknizer

  I've compiled FFMPEG binary from Bambuser project (refer question) and have pushed binary and .so files to application directory.

  But when I try running FFMPEG with input file, it always complains file not found.

  1|root@android:/data/data/com.bambuser.broadcaster # ls                        
  
  cache           libavcodec.so   libavfilter.so  libswscale.so
  
  ffmpeg          libavcore.so    libavformat.so  tutorial.mp4
  
  lib             libavdevice.so  libavutil.so
  
  1|root@android:/data/data/com.bambuser.broadcaster # ./ffmpeg -i tutorial.mp4 out.mp4                                                              
  
  FFmpeg version UNKNOWN, Copyright (c) 2000-2010 the FFmpeg developers
  
    built on May  8 2012 10:11:37 with gcc 4.4.3
  
    configuration: --target-os=linux --cross-prefix=arm-linux-androideabi- --arch=arm --sysroot=/home/tarandeep/tools/android-ndk/platforms/android-3/arch-arm --soname-prefix=/data/data/com.bambuser.broadcaster/lib/ --enable-shared --disable-symver --enable-small --optimization-flags=-O2 --disable-everything --enable-encoder=mpeg2video --enable-encoder=nellymoser --prefix=../build/ffmpeg/armeabi-v7a --extra-cflags='-march=armv7-a -mfloat-abi=softfp' --extra-ldflags=
  
    libavutil     50.34. 0 / 50.34. 0
  
    libavcore      0.16. 0 /  0.16. 0
  
    libavcodec    52.99. 1 / 52.99. 1
  
    libavformat   52.88. 0 / 52.88. 0
  
    libavdevice   52. 2. 2 / 52. 2. 2
  
    libavfilter    1.69. 0 /  1.69. 0
  
    libswscale     0.12. 0 /  0.12. 0
  
  tutorial.mp4: No such file or directory

  I've tried placing file in /sdcard and other locations, I always get same output

• How To Write An Oscilloscope

  29 avril 2012, par Multimedia Mike — General, gme, oscilloscope, visualization

  I’m trying to figure out how to write a software oscilloscope audio visualization. It’s made more frustrating by the knowledge that I am certain that I have accomplished this task before.

  In this context, the oscilloscope is used to draw the time-domain samples of an audio wave form. I have written such a plugin as part of the xine project. However, for that project, I didn’t have to write the full playback pipeline— my plugin was just handed some PCM data and drew some graphical data in response. Now I’m trying to write the entire engine in a standalone program and I’m wondering how to get it just right.

  
  
  

  This is an SDL-based oscilloscope visualizer and audio player for Game Music Emu library. My approach is to have an audio buffer that holds a second of audio (44100 stereo 16-bit samples). The player updates the visualization at 30 frames per second. The o-scope is 512 pixels wide. So, at every 1/30th second interval, the player dips into the audio buffer at position ((frame_number % 30) * 44100 / 30) and takes the first 512 stereo frames for plotting on the graph.

  It seems to be working okay, I guess. The only problem is that the A/V sync seems to be slightly misaligned. I am just wondering if this is the correct approach. Perhaps the player should be performing some slightly more complicated calculation over those (44100/30) audio frames during each update in order to obtain a more accurate graph ? I described my process to an electrical engineer friend of mine and he insisted that I needed to apply something called hysteresis to the output or I would never get accurate A/V sync in this scenario.

  Further, I know that some schools of thought on these matters require that the dots in those graphs be connected, that the scattered points simply won’t do. I guess it’s a stylistic choice.

  Still, I think I have a reasonable, workable approach here. I might just be starting the visualization 1/30th of a second too late.