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• Converting variable bitrate on the fly with ffmpeg

  4 septembre 2016, par Łukasz Bezwerchny

  i do cut some video files with either avidemux or virtualdub using directstream copy, some of the files comes with variable bitrate, the problem is that after such cutting my default video cataloging software shows those files lenght as 0:0, i managed to fix this problem with ffmpeg using "-vcodec copy -acodec copy". The output seem to be fine now but i have another problem, these new files have sound problem, on pot player everything is working fine but on mpc and my cataloging software which also plays media sound gets cut off at the half of the movie, for example 2h movie has only 30min of sound. I did look at the using mediainfo software and it shows me something like : video lenght 1:55, audio lenght 32min. I think the problem is the vbr, i did manage to solve this problem by extracting mp3 file from the troublesome video and convert it to 128kb constant bitrate and again merge the video and new audio file and it seems fine. Video and audio match at the whole video lenght. It’s just a bit of tedious to get things done for a big amount of files to fix, is there a parameter that i could use in one command line to do the job in one process, i mean like :
  ffmpeg -i test.avi -vcodec copy -"convert vbr to 128kb" "save to test2.avi" ?

• bash variable changes in loop with ffmpeg

  17 septembre 2018, par Mike

  I wrote a skript to quickly create short preview clips from vides I recorded on timestamps that I found worth checking out later for cutting.
  My file with the timestamps is written like this

  FILE_NAME1#MM:SS MM:SS
  
  FILE_NAME2#MM:SS MM:SS MM:SS MM:SS

  example :

  MAH01728#02:47 03:34 03:44 05:00 06:08 06:55

  The script looks like this :

  #!/bin/bash
  
  while read f
  
   do
  
  
  
   file=$(echo $f | cut -d"#" -f1)
  
   filename=${file}".MP4"
  
   timestamps=$(echo $f | cut -d"#" -f2)
  
  
  
  for time in $timestamps 
  
   do 
  
    ffmpeg -ss 00:${time}.0 -i "orig/${filename}" -c copy -t 10 "preview/${file}_${time}.MP4"
  
   done
  
  done < $1

  The script gets half of the previews that I want and on the other the filename is messed up and ffmpeg complains that the file is not found :

  orig/714.MP4: No such file or directory
  
  orig/00:58 01:25.MP4: No such file or directory

  So I modified the script for trouble shooting and just put an echo in front of the ffmpeg command - now all file names are correct. What am I missing ?

  ffmpeg -ss 00:01:47.0 -i orig/MAH01714.MP4 -c copy -t 10 preview/MAH01714_01:47.MP4
  
  ffmpeg -ss 00:02:00.0 -i orig/MAH01713.MP4 -c copy -t 10 preview/MAH01713_02:00.MP4
  
  ffmpeg -ss 00:00:58.0 -i orig/MAH01712.MP4 -c copy -t 10 preview/MAH01712_00:58.MP4
  
  ffmpeg -ss 00:01:25.0 -i orig/MAH01712.MP4 -c copy -t 10 preview/MAH01712_01:25.MP4

• How to decode mp3 to pcm by ffmpeg

  30 janvier 2017, par Meph-

  I need decode mp3 audio data to pcm. I have data which starts with mp3 header. Api-example.c doesn’t work, output is strange :

  

  command ffmpeg -i input.mp3 output.wav
  is great, this is what i need. But I cant find way how to do that in code. Does anybody know, where some tutorial with ffmpeg library is ? Thanks

  Edit 2.7.13 :

  Hi again,
  I rebuilt the audio decode example method from ffmpeg and my problem is probably here :

  len = avcodec_decode_audio4(avCodecContext,avFrame, &got_frame,&avPacket);    
  
  int data_size = av_samples_get_buffer_size(NULL,avFrame->channels,avFrame->nb_samples,AV_SAMPLE_FMT_S16P,1);

  data_size is size of data frame from decoder, it depends on number of channels, number of data samples and data type(my data are 16bit PCM stereo encoded to mp3 to 1152 samples of mp3 frame)

  If I open an output file in audacity, correct parameters, which give correct output, are stereo (right), 8bit pcm (wrong) and half sample rate (also wrong), what’s it happened ?

  data before encoding :
  16bit PCM 44100Hz, stereo

  data after decoding :
  8bit PCM 22050Hz, stereo ---> ???!!!

  I’m tired of this....