Recherche avancée

Sur d’autres sites (7096)

• 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.

• decoding of APE (Monkey's Audio) using FFmpeg

  19 juillet 2013, par Imran.vatic

  I am developing a player which plays different audio formats,I have successfully played mp3,acc,wav,ogg,alac,flac formats. But now stuck on APE or Monkey's Audio format, I am using a open source library FFmpeg for decoding.Can someone tell me how to decode APE format,I am successful in creating the packets,storing in a buffer and passing it to FFmpeg in the same fashion as I did for other formats,but FFMpeg is not able to decode those packets.Can someone tell me what wrong I am doing ? What should be the buffer size in case of ape that needs to be passed to ffmpeg.Also header parsing, The decode function of ffmpeg is avcodec_decode_audio3.

• Converted mp4 h264 baseline format loads long time

  10 juillet 2013, par user1830062

  I have converted my video to mp4 x264 baseline format and it works fine with all pc/mobile phones , the problem is it takes long time to load the video while googling came to know that ffmpeg converts and sets the index file at the eof the video so it loads to the end to read and then plays the video, So any advices would be appreciatable to cut short the loading time.
  Note:tryied out QT index swapper2 but dint give much difference , please advice .

  this is the cmd i used to convert -

  ffmpeg -i … -c:v libx264 -profile:v baseline -level 1 …

  Thanks for your time .