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• avcodec/smcenc : improve compression ratio a little

  2 décembre 2022, par Paul B Mahol

  avcodec/smcenc : improve compression ratio a little
  Actually reverts previous change.
  

  • [DH] libavcodec/smcenc.c

• ffmpeg continuous live streaming of dynamic files from directory

  4 septembre 2022, par user72261

  So I have been using ffmpeg to create a live stream from files stored in a folder using concat list.

  


  This works perfectly for local, static content but I have a security system in my warehouse that uploads .mp4 files to my server in the office.

  


  This system doesn't have any live stream functionality so I would like to set up a live stream myself using ffmpeg to continually read the latest files from the folder.

  


  I tried setting up a script to read all files in the directory and sort them based on the file name that is a UNIX timestamp of the date/time the video was taken. Then had the script select the latest segments after that last used segment and create a new concat list such as below

  


  file '00000.mpd'
file '00001.mp4'
file '00002.mp4'
file '00003.mp4'
file '00004.mp4'


  


  After doing some reading I found out to make it loop that file you could reference its self in the list like so

  


  file '00000.mpd'
file '00001.mp4'
file '00002.mp4'
file '00003.mp4'
file '00004.mp4'
file 'concat.txt'


  


  This seemed to work ok until I started getting the error impossible to open concat.txt. This file doesn't exist, which I can only guess is because ffmpeg is trying to open the file as its being updated by the script.

  


  Has anyone come across this before or know another way to do it ?

  


• 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.