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• configure : Remove dcbzl check for e500v1 and e500v2 architectures

  6 février 2014, par Peter Krefting

  configure : Remove dcbzl check for e500v1 and e500v2 architectures
  The DCBZL instruction is not available for the e500v1 and e500v2
  
  architectures, but may still be recognized by the toolchain, so we need to
  
  remove the test for it explicitly for these architectures.
  References : PowerPC™ e500 Core Family Reference Manual (Freescale)
  Found-by : Ståle Kristoffersen <staalebk@ifi.uio.no>
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] configure

• How to put H264 encoded data, inside a container like .mp4 using libAV libraries (ffmpeg) ?

  27 mai 2022, par Vivekanand V

  I have been playing around with the libAV* family of libraries, that comes along with FFMPEG and I am learning how to implement things like encoding, decoding, muxing etc.

  


  I came across this code : https://libav.org/documentation/doxygen/master/encode_video_8c-example.html , that encodes, YUV frames, to a file, using an encoder. In my implementation, I will change the encoder to H264 using avcodec_find_encoder(AV_CODEC_ID_H264) But I do not know, how to put this encoded data, into a proper container (.mp4) using libav libraries, so that the output file, can be played by any media player like VLC, QuickTime, etc...

  


  Can anyone please help me on that ? Every help will be greatly appreciated !

  


• 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.