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• checkasm : updated tests for sw_scale

  13 août 2022, par Swinney, Jonathan

  checkasm : updated tests for sw_scale
  Change the reference to exactly match the C reference in swscale,
  
  instead of exactly matching the x86 SIMD implementations (which
  
  differs slightly). Test with and without SWS_ACCURATE_RND - if this
  
  flag isn't set, the output must match the C reference exactly,
  
  otherwise it is allowed to be off by 2.
  Mark a couple x86 functions as unavailable when SWS_ACCURATE_RND
  
  is set - apparently this discrepancy hasn't been noticed in other
  
  exact tests before.
  Add a test for yuv2plane1.
  Signed-off-by : Jonathan Swinney <jswinney@amazon.com>
  
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libswscale/x86/swscale.c
  • [DH] tests/checkasm/sw_scale.c

• lavc/cbs_av1 : restore CodedBitstreamAV1Context when AVERROR(ENOSPC)

  28 septembre 2022, par Haihao Xiang

  lavc/cbs_av1 : restore CodedBitstreamAV1Context when AVERROR(ENOSPC)
  The current pbc might be small for an obu frame, so a new pbc is
  
  required then parse this obu frame again. Because
  
  CodedBitstreamAV1Context has already been updated for this obu frame, we
  
  need to restore CodedBitstreamAV1Context, otherwise
  
  CodedBitstreamAV1Context doesn't match this obu frame when parsing obu
  
  frame again, e.g. CodedBitstreamAV1Context.order_hint.
  $ ffmpeg -i input.ivf -c:v copy -f null -
  
  [...]
  
  [av1_frame_merge @ 0x558bc3d6f880] ref_order_hint[i] does not match
  
  inferred value : 20, but should be 22.
  
  [av1_frame_merge @ 0x558bc3d6f880] Failed to write unit 1 (type 6).
  
  [av1_frame_merge @ 0x558bc3d6f880] Failed to write packet.
  
  [obu @ 0x558bc3d6e040] av1_frame_merge filter failed to send output
  
  packet
  Reviewed-by : James Almer <jamrial@gmail.com>
  
  Reviewed-by : Wenbin Chen <wenbin.chen@intel.com>
  
  Signed-off-by : Haihao Xiang <haihao.xiang@intel.com>
  

  • [DH] libavcodec/cbs_av1.c

• How to merge two audio files while retaining correct timings with ffmpeg

  31 août 2017, par andy

  Disclaimer : I don’t understand ffmpeg and mostly just copy commands

  I have two webm video files. Their duration is off by about a second.

  I converted these to mp4 and everything was fine :

  ffmpeg -acodec libopus -i 1.webm -r 10 -cpu-used 5 -c:v libx264 -crf 20 -c:a aac -strict experimental -loglevel error 1.mp4

  I then extracted the audio from each video and everything was fine. By fine, I mean, when I playback the audio files by themselves they match the timings on the original video. I used this command to extract the audio :

  fmpeg -i 1.mp4 -map 0:a -vn -acodec copy 1audio.m4a

  I now have two audio tracks and I want to combine them into one, I want to "overlay" them, AND I want to keep the timings the same as they were.

  BUT, whenever I try and combine the audio the timings are off. The "seconds" don’t match the original single audio files.

  I have tried these commands to merge the audio :

  ffmpeg -i 1audio.m4a -i 2audio.m4a -filter_complex "[0:a][1:a]amerge=inputs=2[a]" -map [a] -c:a libfdk_aac mergedaudio.m4a

  ffmpeg -i 2audio.m4a -i 1audio.m4a -filter_complex amix=inputs=2:duration=first mergedaudio.m4a

  ffmpeg -i 1audio.m4a -i 2audio.m4a -filter_complex amerge -ac 2 -c:a libfdk_aac -vbr 4 mergedaudio.m4a

  All the above command have resulted in a merged audio file... but the timings are off.

  What am I doing wrong ?