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• avcodec/h264_parser : Fix undefined left shift

  26 mai 2020, par Andreas Rheinhardt

  avcodec/h264_parser : Fix undefined left shift
  Use an uint32_t for the NAL unit size of an AVC H.264 NAL unit instead
  
  of an int as a left shift of a signed value is undefined behaviour
  
  if the result doesn't fit into the target type.
  Also make the log message never output negative lengths.
  Fixes : left shift of 16711968 by 8 places cannot be represented in type 'int'
  
  Fixes : 36601/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_H264_fuzzer-6581933285965824
  Reviewed-by : Michael Niedermayer <michael@niedermayer.cc>
  
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
  

  • [DH] libavcodec/h264_parser.c

• Input raw audio 512 floating-point values into FFMPEG ?

  29 septembre 2021, par NTPHCM

  In ActionScript 3, I use SoundMixer.computeSpectrum() to takes a snapshot of the current sound wave playing on the stage and places it into the specified ByteArray object.

  


  I want to write the ByteArray into FFMPEG (by std -i pipe:0), but the ByteArray object created is fixed to 512 floating-point values and FFMPEG seem doesn't support that value. The sound in the output is corrupted.

  


  Here's a part of my code in AS3 :

  


  stage.addEventListener(Event.ENTER_FRAME, update);
function update(e:Event):void{
    var soundBA:ByteArray = new ByteArray();
    SoundMixer.computeSpectrum(soundBA, false, 0);
    process.standardInput.writeBytes(soundBA);
}


  


• How to salvage a gigantic, possibly corrupt, AIFF file ?

  9 novembre 2024, par Ross Duncan

  Due to a Sound Track Pro glitch, I have a problematic AIF file. It plays fine in QuickTime Player, and is about 1 hour 50 mins long. However :

  



  

  • It's 3.81GB in size, whereas (I believe) AIF files are only supposed to be 2GB or smaller.

  


  • iTunes refuses to play it.

  


  • Logic Pro gives the error message "party-mix.aif is empty!" when I try to import it.

  


  • VLC will open and play the file, but it ends after an 1 hour (I guess this is the 2GB point).

  


  • Attempting to transcode using ffmpeg to Ogg gives the errors,

    



    


    [aiff @ 0x102051000] max_analyze_duration 5000000 reached at 5001333

    
 


    ....

    
 


    [pcm_s16be @ 0x10205a200] invalid PCM packet

    


    



    but the resulting file plays in VLC up till 1 hour, when it ends.

  


  • Attempting to transcode using ffmpeg to MP3 gives similar results as Ogg, except that the number of channels must reduced from 6 to 2. This is odd because it was a stereo project - where the extra 4 channels came from I have no idea.

  


  • There are a few places where the audio has been very noticeably sped up.

  


  



  My question : how can I transcode this frankenstein monster to MP3 without losing the second half ? I am running Mac OS 10.6.8.