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• Using ffmpeg to convert sound files for use in an android app

  10 janvier 2012, par stefs

  short : i'm trying to simply play a sound file converted with ffmpeg in my android app, but happen to have problems getting it to work.

  long : we have an iphone app and an android app doing the same thing, and i have to port the feature playing a sound on an user interaction. i have the source file in the aiff format, and tried to convert it to mp3 for android. but the app keeps crashing when it tries to load the file

  AssetFileDescriptor fileDescriptor = context.getResources().openRawResourceFd(resid);
  
  final MediaPlayer mp = new MediaPlayer();
  
  mp.setDataSource(fileDescriptor.getFileDescriptor(), fileDescriptor.getStartOffset(), fileDescriptor.getLength());
  
  fileDescriptor.close();
  
  mp.prepare();

  more specifically, mp.setDataSource crashes. some digging around led me to believe that something's wrong with the encoding. the sound file itself resides in res/raw.

  11-29 17:11:48.012: ERROR/SoundManager(15580): java.io.IOException: setDataSourceFD failed.: status=0x80000000
  
  11-29 17:11:48.012: ERROR/SoundManager(15580):     at android.media.MediaPlayer.setDataSource(Native Method)
  
  ...

  what i tried :

  • using a different mp3 that's already used with the same code in a different place. this works.
  • converted it to wav file. this didn't cause the app to crash, but it neither played a sound. that might be a different problem.
  • converted it to ogg ; crashed

  so, the the ffmpeg conversion parameters are as follows :

  $ ffmpeg -i click_24db.aif -f mp3 ~/foobar/wheel_click.mp3
  
  ffmpeg version 0.7.8, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Nov 24 2011 14:31:00 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3)
  
    configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libopenjpeg --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm
  
    libavutil    50. 43. 0 / 50. 43. 0
  
    libavcodec   52.123. 0 / 52.123. 0
  
    libavformat  52.111. 0 / 52.111. 0
  
    libavdevice  52.  5. 0 / 52.  5. 0
  
    libavfilter   1. 80. 0 /  1. 80. 0
  
    libswscale    0. 14. 1 /  0. 14. 1
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  Input #0, aiff, from 'click_24db.aif':
  
    Duration: 00:00:00.01, start: 0.000000, bitrate: 1570 kb/s
  
      Stream #0.0: Audio: pcm_s16be, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Output #0, mp3, to '/Users/xyz/foobar/wheel_click.mp3':
  
    Metadata:
  
      TSSE            : Lavf52.111.0
  
      Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Press [q] to stop, [?] for help
  
  size=       1kB time=00:00:00.05 bitrate=  92.9kbits/s    
  
  video:0kB audio:0kB global headers:0kB muxing overhead 45.563549%

  the resulting file plays nice in itunes, does not play in vlc and crashes when loaded with the android.media.MediaPlayer (note : i first tried it with the SoundPool lib, with both mp3 and ogg, but that didn't work either).

  i also tried the following paramters, which didn't work :

  ffmpeg -i inputfile.aif -f mp3 -acodec libmp3lame -ab 192000 -ar 44100 outputfile.mp3

  i'm working on osx, built ffmpeg with macports today, android api level is 7 (google api, 2.1-update1). looking at the "supported formats" table on dev.android didn't indicate my file to be out of the spec, but i may be mistaken in that.

  i don't have the slightest clue regarding bitrates and so on, so could anybody please point me to the right combination of ffmpeg parameters to get a working mp3 for android ? i don't care if the resulting file would be mp3, ogg or 3gp or whatever.

• How to implement HTTP Live Streaming server on Unix ?

  5 septembre 2012, par alex

  I just realized that Apple required HTTP Live Streaming in order to view videos in iPhone apps. I was not aware of this before... I am now trying to understand what this involves so I can decide whether I want to do the work and make the videos available in 3G or limit video playing to users who are connected to wi-fi.

  I read the overview provided by Apple, and now understand that my server needs to segment and index my media files. I also understand that I don't have to host the content to be able to stream it (I can point to a video hosted somewhere else, right ?).

  What's not clear to me at this point is what to implement on my server (Ubuntu Hardy) to do the actual segmenting and indexing on the fly (once again, I do not host the videos I want to serve).

  I found a link explaining how to install FFmpeg and X264, but I don't know if this is the best solution (since I have an Ubuntu server, I can't use the Apple Live Streaming tools, is it correct ?). Also, I do not understand at which point my server knows that a video needs to be converted and starts the job...

  Any feedback that could help me understand exactly what to do on the server side to be able to stream videos on my iPhone app in 3G would be greatly appreciated ! (Oh, and just it makes any difference, my app back-end is in Rails)

• FFMPEG - Correct command line parameters to create a H.264 Streaming MP4 from a set of images

  5 janvier 2012, par Adam

  I have no problems viewing a video created by my current parameters in a HTML5 browser, Flash player, Windows Phone 7, Android Phone, etc. However, they won't display on iPhone or iPad. If I modify the video using Super and select "Enable Streaming" checkbox they work on iPhone and iPad suddenly. We are suspecting that my command line parameters are incorrect for enabling h.264 streaming and are instead generating a "progressive download" video.

  I'm pretty new to FFMPEG and don't understand a lot of the parameters. Can anyone help correct my existing parameters, or maybe provide the missing parameters, etc ?

  Thanks in advance.

  My current FFMPEG command line paramters are :

  ffmpeg.exe 
   -r 30 
   -threads 4 
   -f image2 
   -i .\frame%05d.jpg 
   -i audioFile  
   -acodec aac 
   -ab 128k 
   -ar 44100
   -vcodec h264 
   -crf 27 
   -coder 1 
   -flags +loop 
   -cmp +chroma 
   -partitions +parti4x4+partp8x8+partb8x8 
   -me hex 
   -subq 5 
   -me_range 16 
   -g 60 
   -keyint_min 25 
   -sc_threshold 40 
   -i_qfactor 0.71 
   -b_strategy 1 
   out.mp4