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• CoreAudio : how to retrieve actual sampling rate of raw data ?

  13 août 2014, par jyavenard

  When attempting to play AAC-HE content in an mp4 container, the reported sampling rate found in the mp4 container appears to be half of the actual sampling rate.

  E.g it appears as 24kHz instead of 48kHz.

  Using the FFmpeg AAC decoder, retrieving the actual sampling rate can be done by simply decoding an audio packet using

  avcodec_decode_audio4

  And looking at AVCodecContext::sample_rate which will be updated appropriately. From that it’s easy to adapt the output.

  With CoreAudio decoder, I would use a AudioConverterRef set the input and output AudioStreamBasicDescription
  and call AudioConverterFillComplexBuffer

  As the converter performs all the required internal conversion including resampling it’s fine. But it plays the content after resampling it to 24kHz (as that’s what the input AudioStreamBasicDescription contains.

  Would there be a way to retrieve the actual sampling rate as found be the decoder (rather than the demuxer) in a similar fashion as one can with FFmpeg ?

  Would prefer to avoid losing audio quality if at all possible, and not downmix data

  Thanks

• avformat/mp3 : large id3 tags break concatenated file detection

  24 avril 2015, par wm4

  avformat/mp3 : large id3 tags break concatenated file detection
  If the file size is much larger than what is indicated in the XING
  
  header, the demuxer assumes it’s a concatenated file, and throws away
  
  the (presumably) incorrect duration information. Unfortunately, this
  
  also triggers if the id3 tags are very large (embedded pictures and
  
  such). Then the half-baked heuristic not only breaks the duration
  
  display, but also gapless audio.
  Fix it by subtracting the size of the headers (the check is off by some
  
  bytes, but that doesn’t matter at all). Note that there could be an
  
  arbitrary amount of tags _after_ the mp3 data, but hopefully these are
  
  not too large to trigger the heuristic in practice.
  Also add a warning when this happens.
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavformat/mp3dec.c

• What ffmpeg filters can I use to achieve shrinking a video and keeping original aspect ratio ?

  3 août 2020, par sivano

  I am trying to do some specific video editing with ffmpeg and am lacking the correct terminology to describe it (and thus read online how to achieve it). Can someone help describe this using standard (ffmpeg/video editing) terminology ?

  


  The below describes it verbosely, and you can see it visually at the end (this was done manually for one image).

  


  I have a video with aspect ratio 1280x720. At some point in the video, I want to shrink the video diagonally (i.e. maintaining aspect ratio) so that it only takes up half the original width (i.e. 640) and then pad what is left with black. The purpose of this is to then overlay some other image on the left hand of the modified video.

  


  Before


  


  After


  


  Thank you