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• lavf/img2dec : Improve detection of valid Quickdraw images.

  29 juin 2015, par Carl Eugen Hoyos

  lavf/img2dec : Improve detection of valid Quickdraw images.
  Detect Quickdraw images without application header with
  
  lower score.
  

  • [DH] libavformat/img2dec.c

• Theatrical quality ffmpeg/x264 encoding of a high-motion 1080p video

  2 décembre 2011, par Ian

  I've been struggling with encoding videos using FFMPEG and x264. The output stutters when played back in Quicktime, while in VLC it shows a lot of compression artifacts at the same places Quicktime stutters. So it seems like Quicktime is stuttering because it's trying to suppress the corruption/artifacts.

  The videos have a lot of random motion in them, including frames where 75% of the pixels will change at a random interval (the video is software generated so it's truly pseudo-random). The compression seems to be choking in these places where it's likely detecting a "scene cut" incorrectly. It also seems to choke at regular intervals where I guess it's doing a keyframe.

  I've based my encoding preset off of the x264-hq preset that comes with FFMPEG. I've tried turning off scene cut detection, and playing with the keyint/g and keyint_min options. Setting g to 1 makes it work, but blows out the filesize. I've tried the lossless presets, but they won't playback at all in Quicktime. Oddly, I haven't had any problems when working with a lower-resolution test video (1440x810).

  Here's the preset I have right now, which works, but yields a file that's approximately 60% larger than the (non-working) hq preset yields. Is there any way to improve upon this ? The filesize doesn't matter much, I just want something that will playback anywhere and be very high quality.

  coder=1
  flags=+loop
  cmp=+chroma
  partitions=+parti8x8+parti4x4+partp8x8+partp4x4+partb8x8
  me_method=umh
  subq=8
  me_range=16
  g=1
  keyint_min=1
  sc_threshold=0
  i_qfactor=0.71
  b_strategy=1crf=20
  qcomp=0.6
  qmin=20
  qmax=51
  qdiff=4
  bf=16
  refs=4
  trellis=1
  flags2=+dct8x8+wpred+bpyramid+mixed_refs
  wpredp=2
  

  Here's the command :

  ffmpeg \
    -r 60 -i "frame-%06d.tiff" \
    -vcodec libx264 -vpre my_preset \
    -threads 0 \
    -r 60 -an -f out.mp4
  

• Mixing a FLV audio stream with a WAV background track, and converting to MP3 with SoX and FFmpeg

  7 septembre 2012, par tubbo

  I'm building a Flash-based recording application for a contracted web site. It streams the recorded voice (via SWF) to a Red5 server, then uses a combination of FFmpeg and SoX to compile the vocal audio with a lower-in-volume background music track. This all has to happen on-demand, that is, when a user "saves" his or her vocal recording.

  Here is an example command I will be running. Names have been changed to protect the innocent. The filenames describe their role in the final file :

  sox --combine mix -p --no-show-progress --norm "|ffmpeg -i /usr/share/red5/webapps/audiorecorder/stream/SPOKEN_VOICE.flv -t wav pipe:1" /var/www/ufiles/music/BACKGROUND_MUSIC.wav - | ffmpeg -i pipe:1 /var/www/ufiles/recordings/COMPILED_AUDIO_RECORDING.mp3

  When I run this command in the shell, this is what happens :

  $ sox --combine mix -p --no-show-progress --norm "|ffmpeg -i audioStream_1321399534128_21.flv -ar 44100 -ac 2 -t wav pipe:1" wrong.wav - | ffmpeg -i pipe:1 ~/www/trauma101.com/compiled.mp3
  
  ffmpeg version N-34884-g7575980, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Nov 15 2011 14:06:49 with gcc 4.4.5
  
    configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libx264 --enable-x11grab --enable-libspeex
  
    libavutil    51. 25. 0 / 51. 25. 0
  
    libavcodec   53. 34. 0 / 53. 34. 0
  
    libavformat  53. 20. 0 / 53. 20. 0
  
    libavdevice  53.  4. 0 / 53.  4. 0
  
    libavfilter   2. 48. 1 /  2. 48. 1
  
    libswscale    2.  1. 0 /  2.  1. 0
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  ffmpeg version N-34884-g7575980, Copyright (c) 2000-2011 the FFmpeg developers
  
    built on Nov 15 2011 14:06:49 with gcc 4.4.5
  
    configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libx264 --enable-x11grab --enable-libspeex
  
    libavutil    51. 25. 0 / 51. 25. 0
  
    libavcodec   53. 34. 0 / 53. 34. 0
  
    libavformat  53. 20. 0 / 53. 20. 0
  
    libavdevice  53.  4. 0 / 53.  4. 0
  
    libavfilter   2. 48. 1 /  2. 48. 1
  
    libswscale    2.  1. 0 /  2.  1. 0
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  [libspeex @ 0x1e36b20] Missing Speex header, assuming defaults.
  
  Input #0, flv, from 'audioStream_1321399534128_21.flv':
  
    Metadata:
  
      novideocodec    : 0
  
      server          : Red5 Server 1.0.0 RC2 Rev: 4295
  
      creationdate    : Tue Nov 15 15:25:41 PST 2011
  
      canSeekToEnd    : true
  
    Duration: 00:00:06.77, start: 0.000000, bitrate: 43 kb/s
  
      Stream #0:0: Audio: speex, 16000 Hz, 1 channels, s16
  
  Invalid duration specification for t: wav
  
  sox FAIL formats: can't open input pipe `|ffmpeg -i audioStream_1321399534128_21.flv -ar 44100 -ac 2 -t wav pipe:1': premature EOF

  I think the issue is stemming from the conversion from FLV to WAV in FFmpeg, and since it's being piped in it causes the whole process to fail. I always get that duration warning, but when FFmpeg outputs to a .wav file and the SoX command is run separately, I can still get a WAV from SoX and convert that to MP3 manually. I'd like to do all this in one line, piping the data between applications.

  What do I do ?