Recherche avancée

Sur d’autres sites (11137)

• ffmpeg merge wav-audio and rtmp-video stream with itsoffset parameter ?

  2 juillet 2015, par whatis01

  I tried to merge WAV audio and RTMP video stream with -itsoffset parameter :

  ffmpeg -i rtmp://server/video/stream_folder/stream_id -itsoffset 00:00:15.0000 -i /audio/1227121.wav \
  
  -codec:v copy -ac 2 -codec:a aac -strict experimental -f flv rtmp://server/sfolder/test_1

  But it works inappropriately (RTMP output is not steady, only voice or only video), and always prompts : mux overhead ratio 2000%, then conversion failed.

• How to create a video from png images using ffmpeg

  28 décembre 2011, par Rajat

  I want to create a video from different png images. My code is :

  ffmpeg -r 20 -f image2 -i slideshow/%d.png -y -s 320x240 -aspect 4:3 out.mp4

  and i receive output :

  FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Sep 27 2011 00:47:07 with gcc 4.1.2 20080704 (Red Hat 4.1.2-50)
  
    configuration: --enable-avfilter --enable-filter=fade
  
    libavutil     50.36. 0 / 50.36. 0
  
    libavcore      0.16. 1 /  0.16. 1
  
    libavcodec    52.108. 0 / 52.108. 0
  
    libavformat   52.93. 0 / 52.93. 0
  
    libavdevice   52. 2. 3 / 52. 2. 3
  
    libavfilter    1.74. 0 /  1.74. 0
  
    libswscale     0.12. 0 /  0.12. 0
  
  Input #0, image2, from 'slideshow/%d.png':
  
    Duration: 00:00:00.25, start: 0.000000, bitrate: N/A
  
      Stream #0.0: Video: png, rgb24, 720x471, 20 fps, 20 tbr, 20 tbn, 20 tbc
  
  [buffer @ 0x9687230] w:720 h:471 pixfmt:rgb24
  
  [scale @ 0x9687600] w:720 h:471 fmt:rgb24 -> w:320 h:240 fmt:yuv420p flags:0xa0000004
  
  Output #0, mp4, to 'out.mp4':
  
    Metadata:
  
      encoder         : Lavf52.93.0
  
      Stream #0.0: Video: mpeg4, yuv420p, 320x240 [PAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 20 tbn, 20 tbc
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Press [q] to stop encoding
  
  Segmentation fault

  What might be the problem ? Please help...
  Currently i am using centos 5 server.

• 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.