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• Allow reading of growing avi files (currently being written)

  5 décembre 2014, par Joakim Plate

  Allow reading of growing avi files (currently being written)
  This uses the RIFF header stored size to figure out the expected AVI
  
  file size, instead of the actual file. To work fully it requires handling
  
  failed avio_seek() instead of assuming they always succeed.
  Some fate file has been cut off and contains half a frame at the end which
  
  previously was not output during demuxing. This frame is now output to
  
  encoder, thus the fate diff update.
  Bug-Id : 261
  
  Signed-off-by : Vittorio Giovara <vittorio.giovara@gmail.com>
  

  • [DBH] libavformat/avidec.c
  • [DBH] tests/ref/fate/aasc
  • [DBH] tests/ref/fate/cllc-argb
  • [DBH] tests/ref/fate/cllc-rgb
  • [DBH] tests/ref/fate/cllc-yuy2-noblock
  • [DBH] tests/ref/fate/cvid-partial
  • [DBH] tests/ref/fate/fic-avi
  • [DBH] tests/ref/fate/truemotion1-24
  • [DBH] tests/ref/fate/vp5
  • [DBH] tests/ref/fate/zmbv-8bit

• mov : Fix handling of zero-length metadata values

  15 décembre 2014, par Martin Storsjö

  mov : Fix handling of zero-length metadata values
  Since 3cec81f4d4, a zero-length metadata value would try to
  
  allocate 2*0 bytes, where av_malloc() returns NULL.
  Always add one to the allocated length, to allow space for
  
  a null terminator in the zero-length case.
  Incidentally, this fixes fate-alac on RVCT 4.0, where a compiler
  
  bug seems to mess up the mov muxer to the point that it writes
  
  the wrong sort of metadata. Previously this bug was undetected,
  
  but since 3cec81f4d4 such mov files started returning
  
  AVERROR(ENOMEM) in the mov demuxer.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] libavformat/mov.c

• Converting the sample rate of alac files with ffmpeg [on hold]

  23 décembre 2014, par sodiumnitrate

  I am using ffmpeg in terminal in OS X Yosemite. I have alac files I have previously converted from .flac without realizing that the sample rate is too high for my iPod. Thus, I have a bunch of .m4a files that I am trying to reduce the sample rate from 88,200 to 44,100. I tried the following, which I modified from a command I found elsewhere :

  for f in *.m4a; do ffmpeg -i "$f" -ar 44100  -y "${f%.m4a}.m4a"; done

  This gives me the following message :

  buffer @ 0x7ffed8e002a0] Unable to parse option value "-1" as pixel format
  
  Last message repeated 1 times
  
  [buffer @ 0x7ffed8e002a0] Error setting option pix_fmt to value -1.
  
  [graph 0 input from stream 0:0 @ 0x7ffed8e003a0] Error applying options to the filter.
  
  Error opening filters!

  And overwrote the files with gibberish.