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• How does ffprobe determine duration ?

  10 janvier 2019, par jasongullickson

  I’m using ffprobe to analyze media files stored on a remote server. This seems to work well, but for some files the duration is missing or incorrect (usually longer than it should be). Other times it returns this information accurately, and it doesn’t seem to be related to the media type (codec, etc.).

  Here’s an example of a command that works :

  ffprobe -v quiet -print_format json -show_streams -show_format http://host.com/file.aiff

  {
  
  "streams": [
  
      {
  
          "index": 0,
  
          "codec_name": "pcm_s16be",
  
          "codec_long_name": "PCM signed 16-bit big-endian",
  
          "codec_type": "audio",
  
          "codec_time_base": "1/44100",
  
          "codec_tag_string": "[0][0][0][0]",
  
          "codec_tag": "0x0000",
  
          "sample_fmt": "s16",
  
          "sample_rate": "44100",
  
          "channels": 2,
  
          "bits_per_sample": 16,
  
          "r_frame_rate": "0/0",
  
          "avg_frame_rate": "0/0",
  
          "time_base": "1/44100",
  
          "start_pts": 0,
  
          "start_time": "0.000000",
  
          "duration_ts": 8494248,
  
          "duration": "192.613333",
  
          "bit_rate": "1411200",
  
          "nb_frames": "8494248",
  
          "disposition": {
  
              "default": 0,
  
              "dub": 0,
  
              "original": 0,
  
              "comment": 0,
  
              "lyrics": 0,
  
              "karaoke": 0,
  
              "forced": 0,
  
              "hearing_impaired": 0,
  
              "visual_impaired": 0,
  
              "clean_effects": 0,
  
              "attached_pic": 0
  
          }
  
      }
  
  ],
  
  "format": {
  
      "filename": "http://host.com/file.aiff",
  
      "nb_streams": 1,
  
      "nb_programs": 0,
  
      "format_name": "aiff",
  
      "format_long_name": "Audio IFF",
  
      "start_time": "0.000000",
  
      "duration": "192.613333",
  
      "probe_score": 100
  
  }
  
  }

  Here’s an example of one that doesn’t :

  ffprobe -v quiet -print_format json -show_streams -show_format "http://host.com/file.wav"

  Which generates this result :

  {
  
  "streams": [
  
      {
  
          "index": 0,
  
          "codec_name": "pcm_s16le",
  
          "codec_long_name": "PCM signed 16-bit little-endian",
  
          "codec_type": "audio",
  
          "codec_time_base": "1/44100",
  
          "codec_tag_string": "[1][0][0][0]",
  
          "codec_tag": "0x0001",
  
          "sample_fmt": "s16",
  
          "sample_rate": "44100",
  
          "channels": 2,
  
          "bits_per_sample": 16,
  
          "r_frame_rate": "0/0",
  
          "avg_frame_rate": "0/0",
  
          "time_base": "1/44100",
  
          "bit_rate": "1411200",
  
          "disposition": {
  
              "default": 0,
  
              "dub": 0,
  
              "original": 0,
  
              "comment": 0,
  
              "lyrics": 0,
  
              "karaoke": 0,
  
              "forced": 0,
  
              "hearing_impaired": 0,
  
              "visual_impaired": 0,
  
              "clean_effects": 0,
  
              "attached_pic": 0
  
          }
  
      }
  
  ],
  
  "format": {
  
      "filename": "http://host.com/file.wav",
  
      "nb_streams": 1,
  
      "nb_programs": 0,
  
      "format_name": "wav",
  
      "format_long_name": "WAV / WAVE (Waveform Audio)",
  
      "bit_rate": "1411200",
  
      "probe_score": 99
  
  }
  
  }

  These two examples are different formats, but I’ve seen it work and not work when the format is the same, I just don’t have an example handy.

  What I’d like to know is if there is something I can change about the parameters I’m using with ffprobe to allow the duration to be determined consistently and accurate, or any information I can find as to how ffprobe works so I figure out how I might change the input files, etc. so they work correctly.

  Alternatively, if there is a different tool that works more reliably (it would need to be an open-source Linux tool) any suggestions or recommendations are welcome.

• FFMPEG loses frames when transcoding segmented video

  17 février 2019, par Vasilis

  When I transcode videos that I have segmented with FFMPEG, it drops consistently 2 frames in the second and third segments.

  The command I use to segment is

  ffmpeg -i input.mp4 -acodec copy -f segment -segment_time 5 -vcodec copy -reset_timestamps 1 -map 0 -y segments/segment%d.mp4

  Then I transcode each file using

  ffmpeg -i segment1.mp4 -c:v libx264 output.mp4

  The segment isn’t missing any frame before transcoding, and when I transcode the non-segmented video it also has the correct frame count.

  I use ffmpeg version 3.2-static on Ubuntu 16.04.5, but I also tried in other Linux distributions with FFMPEG version 4 and still has the same issues.

• ffmpeg audio - video sync issue (audio ahead of video) - while screen recording using x11grab

  12 juillet 2023, par maheshg

  While screen-recording using the below ffmpeg options I consistently get audio ahead of video, delay is in the order of few seconds

  


  ffmpeg cmd :

  


  ffmpeg -y -f x11grab -thread_queue_size 1024 -draw_mouse 0 -video_size 1920x1080 -i :0 -f pulse -thread_queue_size 1024 -i default -c:v libx264 -threads 0 -preset faster -c:a flac -async 1 -vsync 1 -crf 30 -crf_max 33 -f matroska output.mkv

  


  ffprobe output below :

  


  $ ffprobe demo.mkv 
ffprobe version 3.4.4-1~16.04.york0 Copyright (c) 2007-2018 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.10) 20160609
  configuration: --prefix=/usr --extra-version='1~16.04.york0' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Input #0, matroska,webm, from 'demo.mkv':
  Metadata:
    ENCODER         : Lavf57.71.100
  Duration: 01:00:31.93, start: 0.000000, bitrate: 416 kb/s
    Stream #0:0: Video: h264 (High), yuv420p(progressive), 1920x1080, 29.97 fps, 29.97 tbr, 1k tbn, 59.94 tbc (default)
    Metadata:
      ENCODER         : Lavc57.89.100 libx264
      DURATION        : 01:00:31.928000000
    Stream #0:1: Audio: flac, 48000 Hz, stereo, s16 (default)
    Metadata:
      ENCODER         : Lavc57.89.100 flac
      DURATION        : 01:00:30.912000000


  


  I have even tried using avoid_negative_ts flag, but it didnt help with the sync issue.

  


  


  OS : Ubuntu-18.04

  


  ffmpeg version : 3.4

  


  


  Update : (with ffmpeg-4.0)
I have since split this process into 2 parts and tried as shown below (quality is better, but audio/video sync is still an issue)

  


  # STEP-1: screen recording
/usr/bin/ffmpeg -y -f x11grab -thread_queue_size 1024 -draw_mouse 0 \
-video_size 1920x1080 -i :91141925 -f pulse -thread_queue_size 1024 \
-i virtual_sink.monitor -c:v libx264rgb -threads 0 -preset ultrafast \
-c:a flac -ac 1 -crf 0 -f matroska output.mkv

# STEP-2: audio/video encoding
/usr/bin/ffmpeg -y -i output.mkv -c:v libx264 -threads 0 \
-preset faster -pix_fmt yuv420p -c:a copy -ac 1 -crf 25 \
-f matroska final_output.mkv


  


  STEP-1 uses much less CPU during screen recording, but audio is still ahead of video. the display I am using is xvfb one in STEP-1 (since, this is a headless machine in the cloud)

  


  Also, i have tried flags -filter_complex aresample=44100 -vsync 1 in STEP-1 to no avail.

  


  Can someone please help !

  


  


  Update 2 : (with latest ffmpeg from git master)

  


  IT WORKED ! THANKS @llogan for the helpful comments.