Recherche avancée

Sur d’autres sites (6163)

• How to convert an mkv (with subtitles) to something Nexus One friendly ?

  21 décembre 2011, par Daniel Quinn

  I have this ffmpeg one-liner that's been good for generating video files for my Nexus One :

  ffmpeg -i infile.mkv -acodec aac -s 572x238 -vcodec libx264 -vpre ipod640 -ab 128k -b 512k -f mp4 -strict experimental outfile.mp4

  But it does this ignorant of the subtitles in infile.mkv — usually not a problem, unless I'm dealing with a non-english movie. In cases like this, I'd like to use the Japanese audio track, and the English subtitles.

  The funny bit is that I can use mplayer to play it using -alang and -slang, but don't know how to use mencoder to make Nexus One friendly videos. I can use ffmpeg to generate Nexus One friendly videos, but can't figure out how to get it to use a specific subtitle track.

  If someone can solve one of these for me, I'll be a happy camper.

• SOLVED : ffmpeg keeps converting second ac3 audio stream to mp2

  27 janvier 2015, par user1121445

  I’m new to this forum, and hope someone can help.
  I am trying to add a second audio track (chinese.ac3) to an XVID video (vts_01.avi) that already has an ac3 track.

  These audio tracks are encoded to ac3, 48000 Hz, stereo, 128 kb/s, and I would like to keep them that way — just multiplex the streams without transcoding.

  This is the command I am using :

  ffmpeg -i vts_01.avi -vcodec copy -i Chinese.ac3 -acodec copy -map 0:0 -map 0:1 -map 1:0 muxed2.avi -newaudio

  ffmpeg does its work, except for the fact that it converts the second track to mp2, 64 kb/s.

  Here is a relevant excerpt from the output, where the key part is ’Stream #0.2 : Audio : mp2, 48000 Hz, stereo, s16, 64 kb/s’ where I would like ’Stream #0.2 : Audio : ac3, 48000 Hz, stereo, 128 kb/s’ :

  Input #0, avi, from 'vts_01.avi':
  
    Metadata:
  
      encoder         : Lavf53.5.0
  
    Duration: 02:03:26.40, start: 0.000000, bitrate: 1954 kb/s
  
      Stream #0.0: Video: mpeg4, yuv420p, 720x352 [PAR 1:1 DAR 45:22], 23.98 tbr,
  
  23.98 tbn, 23.98 tbc
  
      Stream #0.1: Audio: ac3, 48000 Hz, stereo, s16, 128 kb/s
  
  [ac3 @ 018A7440] max_analyze_duration 5000000 reached at 5024000
  
  [ac3 @ 018A7440] Estimating duration from bitrate, this may be inaccurate
  
  Input #1, ac3, from 'Chinese.ac3':
  
    Duration: 02:03:26.36, start: 0.000000, bitrate: 128 kb/s
  
      Stream #1.0: Audio: ac3, 48000 Hz, stereo, s16, 128 kb/s
  
  File 'muxed2.avi' already exists. Overwrite ? [y/N] y
  
  Output #0, avi, to 'muxed2.avi':
  
    Metadata:
  
      ISFT            : Lavf53.5.0
  
      Stream #0.0: Video: mpeg4, yuv420p, 720x352 [PAR 1:1 DAR 45:22], q=2-31, 23.
  
  98 tbn, 23.98 tbc
  
      Stream #0.1: Audio: ac3, 48000 Hz, stereo, 128 kb/s
  
      Stream #0.2: Audio: mp2, 48000 Hz, stereo, s16, 64 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
    Stream #0.1 -> #0.1
  
    Stream #1.0 -> #0.2

  I have tried a number of different combinations, including explicitly forcing the stream to ac3 with :

  ffmpeg -i vts_01.avi -vcodec copy -i chinese.ac3 -acodec ac3 -ac 2 -ar 48000 -ab 128k -map 0:0 -map 0:1 -map 1:0 muxed2.avi -newaudio

  Same result.

  I have also tried to assign a codec to the stream with stream specifier(based on http://ffmpeg.org/ffmpeg.html#toc-Stream-specifiers-1, but these options are not recognized by my ffmpeg.

  I am running out of things to try. Any help will be appreciated.

• Can't Stream Ogg From ffmpeg Through stdout

  16 janvier 2012, par dave mankoff

  To get the the crux of it, why does the first command work, but the second command does not. They produce slightly differently sized files with different contents :

  ffmpeg -i test.wav -f ogg -acodec libvorbis test.a.ogg
  
  
  
  ffmpeg -i test.wav -f ogg -acodec libvorbis - > test.b.ogg

  test.a.ogg will play properly and has no problems. test.b.ogg starts in the middle of the source audio and has stops and gaps in the audio. It also does not report the length of the track.

  I want to transcode source files on the fly into ogg for a program I am writing and I am trying to pipe the stdout from ffmpeg into my program. Putting the results into an intermediary file will kill performance since the transcoding is supposed to happen on demand.