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• FFMPEG Batch Copy Metadata from "Folder1File1.mp3" to "Folder2File1.mp3" in different folders

  27 février 2020, par Vektorz

  I have two separate files in separate folders with the same name and I would like to transfer the metadata from the file in "folder1" to "folder2".
  Then I would like to add a whole bunch of files fitting this same format and batch transfer all of the metadata information.

  From a stack exchange thread I’ve tried :

  "The following script will loop through the the files in one directory, find corresponding files in a second directory and then combine these two files into a third output directory
  
  
  
  dir1=FIRST DIRECTORY
  
  dir2=SECOND DIRECTORY
  
  output=OUTPUT DIRECTORY
  
  for file in $(ls $dir1); do
  
    ffmpeg -i "$dir1/$file" -i "$dir2/$file" -map 1 -c copy \
  
     # copies all global metadata from in0.mkv to out.mkv  
  
     -map_metadata 0 \
  
     # copies video stream metadata from in0.mkv to out.mkv
  
     -map_metadata:s:v 0:s:v \
  
     # copies audio stream metadata from in0.mkv to out.mkv
  
     -map_metadata:s:a 0:s:a \
  
     "$outdir/$file"
  
  done"

  But for the life of me I cannot get this to work properly and it is a bit overkill. He continues on saying :

  If you want to make something reuseable you could put this in a script with the following header (remove the assignment for dir1, dir2 and output in the script above). And then call it as script.sh dir1 dir2 outdir
  
  
  
  #!/bin/bash
  
  set -x errexit # exit immediately on error
  
  dir1="$1"
  
  dir2="$2"
  
  output="$3"

  And I am totally lost. Can someone please help me to get this to work and walk me through it a bit easier as I’m fairly inexperienced with code/FFMPEG.

  Thank you.

• FFmpeg playing audio slowly after conversion from AAC

  18 janvier 2014, par AnthonyM

  I'm attempting to convert an AAC audio stream for playback. I've discovered that I need to convert from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16 but when I do so the audio plays back at about half speed.

  swr = swr_alloc();
  
  assert(av_opt_set_int(swr, "in_channel_layout", audioContext->channel_layout, 0) == 0);
  
  assert(av_opt_set_int(swr, "out_channel_layout", audioContext->channel_layout, 0) == 0);
  
  assert(av_opt_set_int(swr, "in_sample_rate", audioContext->sample_rate, 0) == 0);
  
  assert(av_opt_set_int(swr, "out_sample_rate", 44100, 0) == 0);
  
  assert(av_opt_set_int(swr, "in_sample_fmt", audioContext->sample_fmt, 0) == 0);
  
  assert(av_opt_set_int(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0) == 0);
  
  swr_init(swr);

  There is my code to convert. The input sample rate is 44100 and the audio is stereo.

  I call the code with

  swr_convert(swr, &output, aDecodedFrame->nb_samples, (const uint8_t**)aDecodedFrame->extended_data, aDecodedFrame->nb_samples) >= 0)

• arm : vp9itxfm : Reorder iadst16 coeffs

  31 décembre 2016, par Martin Storsjö

  arm : vp9itxfm : Reorder iadst16 coeffs
  This matches the order they are in the 16 bpp version.
  There they are in this order, to make sure we access them in the
  
  same order they are declared, easing loading only half of the
  
  coefficients at a time.
  This makes the 8 bpp version match the 16 bpp version better.
  This is cherrypicked from libav commit
  
  08074c092d8c97d71c5986e5325e97ffc956119d.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libavcodec/arm/vp9itxfm_neon.S