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• ffmpeg throws error when trying to remove subtitle track only when artwork is present [on hold]

  2 juillet 2014, par user226372

  Hi ffmpeg users/gurus/developers,

  I am having a peculiar problem.

  I have a media with 5 tracks (Notice the fifth one is album artwork)

  Stream #0:0(und) : Video : h264 (High)
  Stream #0:1(und) : Audio : aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 164 kb/s (default)
  Stream #0:2(und) : Audio : ac3 (ac-3 / 0x332D6361), 44100 Hz, stereo, fltp, 224 kb/s
  Stream #0:3(eng) : Subtitle : mov_text (tx3g / 0x67337874), 1280x60, 0 kb/s (default)
  Stream #0:4 : Video : mjpeg, yuvj420p(pc), 1000x1500 [SAR 1:1 DAR 2:3], 90k tbr, 90k tbn, 90k tbc

  All i am trying is to remove the subtitle track Stream #0:3(eng) : Subtitle by this command

  ffmpeg -i input.mkv -map 0:0 -map 0:1 -map 0:2 -map 0:4 -c copy output.mkv

  as explained here

  https://trac.ffmpeg.org/wiki/How%20to%20use%20-map%20option

  But i am getting this error

  Stream mapping:
  
  Stream #0:0 -> #0:0 (copy)
  
  Stream #0:1 -> #0:1 (copy)
  
  Stream #0:2 -> #0:2 (copy)
  
  Stream #0:4 -> #0:3 (copy)
  
  [ipod @ 0x103011000] Could not find tag for codec mjpeg in stream #3, codec not currently supported in container
  
  Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument

  Conversion failed !

  If the input media doesn’t contain an artwork this error is not happening

  Any ffmpeg gurus who can help me with this issue please !!!!

• How to convert MP4 to MP3 in VLC using Command-Prompt ?

  9 juillet 2016, par user6472523

  I have close to 0 programming experience but would like to figure out how to do this and have been trying to figure out how to do this for awhile. I’m getting very frustrated - wish I could understand it. I’m trying to convert a bunch of MP4 files to MP3 files. I would like to keep the original file and just spit out a bunch of MP3 files with same name as the original file.

  From here : https://wiki.videolan.org/How_to_Batch_Encode/

  Apparently, all you need to do is use this code in the Command-Prompt. I have never really used the Command-Prompt before so no idea what to do.

  FOR /R %%G IN (*.m4a) DO (CALL :SUB_VLC "%%G")
  
  FOR /R %%G IN (*.m4a.mp*) DO (CALL :SUB_RENAME "%%G")
  
  GOTO :eof
  
  
  
  :SUB_VLC
  
   SET _firstbit=%1
  
   SET _qt="
  
   CALL SET _newnm=%%_firstbit:%_qt%=%%
  
   SET _commanm=%_newnm:,=_COMMA_%
  
   REM echo %_commanm%
  
   CALL "C:\Program Files\VideoLAN\VLC\vlc" -I dummy -vvv %1 --sout=#transcode{acodec="mpga",ab="512","channels=2",samplerate="44100"}:standard{access="file",mux="mpeg1",dst="%_commanm%.mp3"} vlc://quit
  
  GOTO :eof
  
  
  
  :SUB_RENAME
  
   SET _origfnm=%1
  
   SET _endbit=%_origfnm:*.m4a=%
  
   CALL SET _newfilenm=%%_origfnm:.m4a%_endbit%=.mp3%%
  
   SET _newfilenm=%_newfilenm:_COMMA_=,%
  
   COPY %1 %_newfilenm%
  
   DEL %1
  
  GOTO :eof
  
  
  
  :eof

  1. So far I changed the directory in the Command-Prompt to the
     folder where the MP4 files are located.
  2. In the code, I’ve changed all mention of .m4a to .mp4.
  3. I’ve change mpga to mp3.
  4. I’ve changed "C :\Program Files\VideoLAN\VLC\vlc" to where my vlc program
     is located.

  What else do I need to do ? And what do all the %% mean in the code ?

  Thank you so much !

• FFMpeg + OpenAL

  24 juillet 2013, par Dekowta

  I've been having a problem decoding different codecs into PCM 16 little-endian which is the supported format for OpenAL (I believe that is correct).

  I am using the windows pre-compiled libraries for FFMpeg (dont know if that effects anything).

  The code I have so far is here http://pastebin.com/T4wGL72a

  I have aiff and wav working (though with single channel OpenAL wants to have half the frequency and im not really sure why).

  The main problem is encoded audio files.

  mp3 files play sometimes (sometime get missing header) but end up stuttering with noise and are high pitched.

  ac3 files played the first preloaded buffers and then the frame would not have any more decoded data in the data array (did not check to see if the packet fetched encoded data or not).

  ogg files are the same as ac3

  m4a files would never finish the frame so I have no idea what happens when they do.

  wma files stutter and have a lot of noise but it is not high pitched like mp3 files

  I don't work with audio very often so its difficult for me to get my head round it and this is also the first time I've worked with FFMpeg.

  I'm sure there is something i'm doing very wrong I just need a hand on pointing out where it is.

  Thanks

  Chris.