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• fftools/ffmpeg : remove a cleanup block at the end of transcode()

  14 octobre 2022, par Anton Khirnov

  fftools/ffmpeg : remove a cleanup block at the end of transcode()
  Some of it is already duplicated in ost_free() - those parts can be just
  
  dropped. The rest is moved to ost_free(), as it properly belongs there.
  

  • [DH] fftools/ffmpeg.c

• avformat/http: split the fragment part from the HTTP URL before the request

  3 février 2020, par Marton Balint

  avformat/http: split the fragment part from the HTTP URL before the request
  RFC 3986 states that the fragment identifier is separated from the rest of the
  
  URI prior to a dereference, and thus the identifying information within the
  
  fragment itself is dereferenced solely by the user agent.
  Signed-off-by : Marton Balint <cus@passwd.hu>
  

  • [DH] libavformat/http.c

• How to insert frames to compensate for frames lost during capture

  5 novembre 2015, par JMor

  My original clip was 22:47 long. I captured the video in avi with Ut Video Lossless Codec at 29.97 fps, with pcm 16 bit unsigned audio.
  I am using Virtualdub with VHScrCap driver for capture.
  Virtualdub and mpc and potplayer play the captured file apparently too fast, but with the right audio pitch in the first 3-4 min, but high pitch in the rest of the video. The duration is 19:06, shorter than the original 22:47 (confirmed by mediainfo)
  The cause of the problem seems to be that I am losing more frames when capturing large HD frames.

  Regular encoding

  Encoding captured clip to mp4 :

  ffmpeg -ss 3.25 -i input.avi -map 0:0 -map 0:1 -threads 0 -c:v libx264 -profile:v main \
  
  -preset:v medium -level 3.1 -x264opts crf=26.0 -aspect 16:9 -t 1112.69 \
  
  -y -f mp4 -vf "crop=1432:808:4:46, hqdn3d=1.5:1.5:6:6, \
  
  scale=1216:684, pad=1280:720:32:18" -c:a ac3 -ac 2 -ar 48000 -b:a 160k \
  
  output.mp4

  The output is 18:32 long, framerate is still 29:97. The audio pitch is OK in the first 2 minutes, and way too high in the rest of the video.

  Trying to correct

  I try to correct it in three steps by (1) encoding a video stream that is slowed down to 23.976 fps and extracting a wav audio stream, (2) slowing speed and pitch of audio and (3) remuxing video and audio :
  (1)

  ffmpeg -ss 3.25 -i input.avi -threads 0 \
  
  -c:v libx264 -profile:v main -preset:v medium -level 3.1 -x264opts crf=26.0 \
  
  -aspect 16:9 -t 1390.862 -an -y -f mp4 -r 24000/1001 \
  
  -vf "crop=1432:808:4:46, hqdn3d=1.5:1.5:6:6, scale=1216:684, pad=1280:720:32:18, \
  
  setpts=1.25*PTS" video_out.mp4  \
  
  -t 1112.69 -y -vn -f wav  audio_out.wav

  (2) The wav audio stream is then slowed down with lower pitch with sox :

  sox --norm audio_out.mp4.wav audio_out-24.wav speed 0.8

  (3) The two streams are then remuxed with :

  ffmpeg -i video_out.mp4 -i audio_out-24.wav -map 0:0 -map 1:0 -c:v copy \
  
  -c:a ac3 -ac 2 -af aresample=resampler=soxr -ar 48000 -b:a 160k \
  
  final_output.mp4

  This time, the video duration (23:10) is closer to the original, the pitch is OK for the whole video except for the first 2-3 minutes, where it is (predictably) too low.

  I have a sense that (1) the capture log, and ffprobe give the frame by frame information that show what is the ’instantaneous’ real frame rate, and (2) that information is not used by ffmpeg encoding, but presumably could be used to correct the frame rate by inserting duplicate or interpolated frames to restitute the correct frame rate. I suspect I could get the information from (1), but have no clue how to do (2).

  If someone familiar with this type of issue could give me some advice, and point me in the right direction, I would really appreciate.