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• FFmpeg inaccurate outputs [closed]

  27 juillet 2012, par user1557780

  Possible Duplicate :
  ffmpeg : videos before and after conversion aren't the same length

  Recently, I've been trying to use FFmpeg for an application which requires a VERY accurate manipulation when it comes to the time parameter (milliseconds resolution). Unfortunately, I was surprised to find out that FFmpeg's manipulation functionalities return some inaccurate results.

  Here is the output of 'ffmpeg' :

  ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Jul 25 2012 19:55:05 with gcc 4.2.1 (Apple Inc. build 5664)
  
    configuration: --enable-gpl --enable-shared --enable-pthreads --enable-libx264 --enable-libmp3lame
  
    libavutil      51. 54.100 / 51. 54.100
  
    libavcodec     54. 23.100 / 54. 23.100
  
    libavformat    54.  6.100 / 54.  6.100
  
    libavdevice    54.  0.100 / 54.  0.100
  
    libavfilter     2. 77.100 /  2. 77.100
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0. 15.100 /  0. 15.100
  
    libpostproc    52.  0.100 / 52.  0.100

  Now, let's assume I want to rip the audio track of 'foo.mov'. Here is the relevant output of 'ffmpeg -i foo.mov' :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'foo.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      creation_time   : 2012-07-24 23:16:08
  
    Duration: 00:00:40.38, start: 0.000000, bitrate: 805 kb/s
  
      Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 480x360, 733 kb/s, 24.46 fps, 29.97 tbr, 600 tbn, 1200 tbc
  
      Metadata:
  
        rotate          : 90
  
        creation_time   : 2012-07-24 23:16:08
  
        handler_name    : Core Media Data Handler
  
      Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, s16, 63 kb/s
  
      Metadata:
  
        creation_time   : 2012-07-24 23:16:08
  
        handler_name    : Core Media Data Handler

  As you probably noticed, the video file duration is 00:00:40.38. Using the following command, I ripped it's audio track :

  'ffmpeg -i foo.mov foo.wav'

  Output :

  Output #0, wav, to 'foo.wav':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      creation_time   : 2012-07-24 23:16:08
  
      encoder         : Lavf54.6.100
  
      Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
  
      Metadata:
  
        creation_time   : 2012-07-24 23:16:08
  
        handler_name    : Core Media Data Handler
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (aac -> pcm_s16le)
  
  Press [q] to stop, [?] for help
  
  size=3482kB time=00:00:40.42 bitrate= 705.6kbits/s    
  
  video:0kB audio:3482kB global headers:0kB muxing overhead 0.001290%

  As you can see, the output file is longer than the file in the input.

  Another example is audio (and video) file trimming :
  Let's assume I would like to use ffmpeg for audio file trimming. I used the next command :

  'ffmpeg -t 00:00:10.000 -i foo.wav trimmed_foo.wav -ss 00:00:25.000'

  Output :

  [wav @ 0x10180e800] max_analyze_duration 5000000 reached at 5015510
  
  Guessed Channel Layout for  Input Stream #0.0 : mono
  
  Input #0, wav, from 'foo.wav':
  
    Duration: 00:00:40.42, bitrate: 705 kb/s
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
  
  Output #0, wav, to 'trimmed_foo.wav':
  
    Metadata:
  
      encoder         : Lavf54.6.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
  
      Press [q] to stop, [?] for help
  
  size=864kB time=00:00:10.03 bitrate= 705.6kbits/s    
  
  video:0kB audio:864kB global headers:0kB muxing overhead 0.005199%

  Again, the output file is 30 milliseconds longer than I expected.

  I tried, for a long time, to research the issue without any success. When I use audacity for the same functionality, it does it very accurately !

  Does anyone have any idea how to solve this problem ?

• Mp4 to dynamic adaptive hls with multiple bitrates using ffmpeg

  10 décembre 2023, par Awais fiaz

  I tried converting mp4 video to HLS for online streaming which I have successfully done using FFmpeg.

  



  Command :

  



  ffmpeg -i /var/www/html/file_conversion/heli.mp4 -map 0 -profile:v baseline -level 3.0 -s 640x360 -c:v libx264 -b:v 500k -c:a libfdk_aac -b:a 320k -hls_list_size 0 -start_number 0 -hls_init_time 0 -hls_time 2  -f hls /var/www/html/file_conversion/hlstest2/heli.m3u8


  



  But now I am trying to convert the same video with multiple bitrates for dynamic adaptive streaming.

  



  Any idea how can I achieve this ?

  


• create mono and stereo stream in the same file with ffmpeg ?

  24 juillet 2017, par user2216280

  I have a film with 2 mono audio stream of avi file.

  I would like to create a third file, with one stereo stream from the 2 mono stream, and one mono stream, addition of the 2 monos audio source...
  here’s the code to make mono stream :

  ffmpeg -i input.avi-ac 1 mono.avi

  here’s the code to make an stereo stream from 2 mono stream :

  ffmpeg -i input.avi-filter_complex "[0:1][0:2] amerge=inputs=2" -c:a pcm_s16le output.avi

  How could I merge those 2codes to make one audio file with one stero track, and one mono track ?
  tiouss ! thanks in advance