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• Creating video with multiple audios and one still image with ffmpeg [on hold]

  4 mai 2014, par user3601509

  Hi i want to create many videos with multiples audios and one single picture.

  For example i have song1.mp3, song2.mp3, song3.mp3 and one single picture. What i need to do is to create many videos and name that videos as : song1.mp4, song2.mp4, song3.mp4

  I want to create many videos at once. Because i have a huge amount of songs and i need a lot of videos with that songs, sorry for my english.

  I have this script for only one picture and one mp3, but i need to create bulk videos can you help me ? ffmpeg -loop 1 -shortest -r 0.1 -i online.mp3 -i imagen.png -ab 128k output.avi

  Many thanks

• Android code FFMPEG. Help me about using ffmpeg command

  4 mai 2016, par Badman

  I have a videos with large size(22MB). I want resizing and crop that videos to small size (1 2 MB). Example size : 480 x 480.
  This is code i using :
  

  -i " + pathSourceVideos + " -strict experimental -vcodec libx264 -crf 27 -vf crop=out_w=in_h -threads 5 -preset ultrafast -strict -2 " +
  pathVideoComplete ;

  This code too slow.
  I have check time process done :
  With videos 14s : process in 02:44s
  With videos 10s : process in 02:00s
  Please give me some suggest.
  Sorry about my english.

• using pocketsphinx_continuous with a .wav file

  3 avril 2013, par user2242131

  I am attempting to write an application that will allow a user to speak a small set of commands from a remote system and have them executed on my server. Using pocketsphinx to parse the spoken text. When run locally with the microphone, pocketsphinx_continuous works perfectly no matter how I slur the words. But when importing the audio file and using ffmpeg to downsample the audio to a single channel, 16 bit PCM file, it will parse the first word without difficulty. Then it will skip everything else and treat it as . I am confident that the problem is in the file format and not in the pocketsphinx configuration.

  Using command line
  ffmpeg -y -i Sound\AddSheet.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav
  in a batch file.

  The bottom of the output I get is :

  INFO: fsg_search.c(1407): Start node ADD.0:5:47
  
  INFO: fsg_search.c(1407): Start node <sil>.0:2:49
  
  INFO: fsg_search.c(1446): End node <sil>.126:128:305 (-486)
  
  INFO: fsg_search.c(1662): lattice start node <s>.0 end node <sil>.126
  
  INFO: ps_lattice.c(1352): Normalizer P(O) = alpha(<sil>:126:305) = -175371
  
  INFO: ps_lattice.c(1390): Joint P(O,S) = -176076 P(S|O) = -705
  
  000000000: ADD USER
  
  </sil></sil></s></sil></sil>

  Which is not the audio in the file. The words spoken in the file are "ADD SPREADSHEET", which works perfectly from the same microphone without the intervening .wav file.

  I have tried increasing the audio volume and decreasing the background noise using sox :

  sox -v 3.0 Sound\%1 Sound\%1-loud.wav ffmpeg -i Sound\%1-loud.wav -vn -ss 00:00:00 -t 00:00:01 -y Sound\%1-noiseaud.wav 
  
  sox Sound\%1-noiseaud.wav -n noiseprof Sound\%1-noise.prof 
  
  sox Sound\%1 Sound\%1-clean.wav noisered sound\noise.prof 0.21 
  
  ffmpeg -y -i Sound\%1-clean.wav -ac 1 -f s16le -acodec pcm_s16le -ar 16k AddTmp.wav

  with no noticeable effect on the final results.

  If you look at the output you will notice that fsg_search.c has found ADD as the start node, then silence for the remainder. Please help on this.