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• File conversion and deletion on Windows

  22 juillet 2016, par Lyuba B.

  I’ve got some 90 000 .m4a audio recordings, all grouped in various sub-folders depending on their purpose. I want to convert each one to .wav and then delete the original .m4a after successful conversion. For reasons beyond my control I’m doing this on a Windows machine with limited capacity. I need to delete as I go along, because I don’t have the space to keep both versions.

  I have the locations of all files stored, and can easily generate a batch file to do conversion and deletion using ffmpeg and DEL iteratively (which is a solution I’ve seen suggested in several places). But is there a way to check that the conversion has been successful before deleting ? I really don’t fancy losing the data by accident.

• 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.

• Static ffmpeg library windows

  29 novembre 2022, par peter

  Im trying to use the ffmpeg libraries (libavutil, libswscale, ...) in a bigger project by statically linking them, im not interested in the binaries. Now the issue comes down to getting them statically built. I tried searching for pre-compiled builds but without success.

  


  Following the docs, i compiled them on wsl2 for win64 with

  


  ./configure --arch=x86_64 --target-os=mingw32 --cross-prefix=i686-w64-mingw32- --disable-shared --enable-static
make


  


  After that i copied the libavformat/libavformat.a as libavformat.lib (which should be fine since they are cross-compiled ar archives ?) into my project and linked them with CMake.

  


  When compiling smth like

  


  AVFormatContext* pFormatContext = avformat_alloc_context();


  


  i get a linker error saying

  


  lld-link : error : undefined symbol: avformat_alloc_context


  


  When looking at the libs with dumpbin -linkermember libavformat.lib, i can see

  


  >dumpbin -linkermember libavformat.lib
...
  135CD92 _avformat_alloc_context
...


  


  notice the underscore.

  


  Can i not link the cross-compiled *.a files directly into a windows executable ?

  


  Honestly at this point any ideas are welcome.

  


  Thanks