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• FFMPEG - Get the exact calculated audio filesize after encode

  14 janvier 2016, par Youssef El Gharbaoui

  Im trying to guess an audio (mp3) filesize before encode with ffmpeg, afterward, need to have the exact calculated filesize.

  Here is the formula im using to predict and calculate the filesize (hope im not wrong) :

  ( Bitrates x Duration ) / 8) x 1000 = Filesize in Bytes.

  Im going to give a real example so that everyone can understand the use case.

  Example :

  Having an m4a file with the following data :

  • Name : Assuming xxx.m4a
  • Filesize : 8 304 014 bytes (8,3 Mo)
  • Bitrates : 256k
  • Duration : 260 seconds

  Expected filesize : ( (256 x 260) / 8 ) x 1000 = 8 320 000 bytes

  Then im running the following ffmpeg command :

  ffmpeg -i xxx.m4a -f mp3 -y -minrate 256k -maxrate 256k -bufsize 256k -b:a 256k -fs 8320000 output.mp3

  Console output :

  ffmpeg version 2.7.2 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with Apple LLVM version 6.1.0 (clang-602.0.53) (based on LLVM 3.6.0svn)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/2.7.2_1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang     --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-vda
  
    libavutil      54. 27.100 / 54. 27.100
  
    libavcodec     56. 41.100 / 56. 41.100
  
    libavformat    56. 36.100 / 56. 36.100
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 16.101 /  5. 16.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.100 /  1.  2.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'xxx.m4a':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2mp41
  
      encoder         : Lavf56.36.100
  
    Duration: 00:04:20.53, start: 0.000000, bitrate: 254 kb/s
  
      Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 253 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler
  
  Output #0, mp3, to 'output.mp3':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2mp41
  
      TSSE            : Lavf56.36.100
  
      Stream #0:0(und): Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp, 256 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler
  
        encoder         : Lavc56.41.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (aac (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  size=    8127kB time=00:04:20.02 bitrate= 256.1kbits/s    
  
  video:0kB audio:8127kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.011765%    

  Problem and Questions :

  • Can you tell me why im I getting an output with 8 322 546 bytes and
    not 8 320 000 as expected ?
  • Is there something wrong in my formula or the ffmpeg command ?
  • What solution can you suggest to get the exact predicted filesize ?

  Thank you in advance.

• FFMPEG convert flv to mp4 without losing quality [closed]

  9 juillet 2013, par Chad Whitaker

  Currently I'm converting .flv flash files to .webm and .mp4 files to be used with an HTML5 video player.

  _

  I use this ffmpeg command for WEBM : ffmpeg -i filename.flv -vcodec libvpx -acodec libvorbis filename.webm
  which works great and the file size stays relatively the same (imporatant).

  _

  For MP4 I'm using the following : ffmpeg -i filename.flv -sameq -ar 22050 filename.mp4

  It converts to .mp4 fine and prevents any quality loss but the file size almost triples in size.

  What ffmpeg formula could be used to convert to MP4 without losing too much quality and preventing a much higher file size ?

• FFmpeg uses too much memory when repeating split, select, overlay

  13 novembre 2020, par finefoot

  I'm running

  


  ffmpeg -i input.mp4 -filter_complex_script script.txt output.mp4


  


  with the following minimal example script :

  


  split[tmp],
select='between(t,1,2)',
select='between(n,0,1)',
[tmp]overlay=enable='between(t,1,2)':eof_action=repeat,
split[tmp],
select='between(t,3,4)',
select='between(n,0,1)',
[tmp]overlay=enable='between(t,3,4)':eof_action=repeat


  


  What I want to do is to take 1 frame at a certain position and repeat it for a certain duration, basically "pausing" the video, while overwriting to keep the same output length. In the example, I'm doing this twice : I'm using split[tmp] to get a second input stream to work on, select the time at position 00:01 with select='between(t,1,2)', select the first frame from that position with select='between(n,0,1)' and finally overlay that frame over the input. This repeats for a second time at position 00:03. I have tested this and it does exactly what I'm looking for.

  


  However, in my real script, I'm repeating this about 1000 times for different positions in the stream (and for shorter durations than 1 second) which results in running out of memory. What am I doing wrong ? What can I do to optimize ?