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• ffmpeg timestamps on extracted frames wrong / extract frames a equal timestamps

  21 juillet 2021, par Sam

  I have a set of videos with different length, sizes, fps etc. and extract from each video exactly 60 frames as images, with timestamps put on each frame.

  


  For this purpose I divide each videos length by 60 and use fps=1/x to extract the frames at even seconds from the video.

  


  This works fine but the printed timestamps are slightly off.

  


  This is the code I use (part of a bash script).

  


  

  1. Compute intervals :

  


  


  output_frame_count=60
video_duration=$(ffprobe -v error -show_entries format=duration -of default=noprint_wrappers=1:nokey=1 -i file.mp4)
capture_distance=$(bc <<< "scale=0; ($video_duration / $output_frame_count)")


  


  

  2. Then run ffmpeg with the arguments

  


  


  ffmpeg -i file.mp4 -vf "fps=1/$capture_distance,scale=-1:$height,drawtext=fontfile=/usr/...: text='%{pts\:gmtime\:0\:%H\\\\\:%M\\\\\:%S}: x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=1: boxcolor=0x00000000@1" frames%04d.png


  


  I checked the images and all timestamps are wrong, compared to the original image from which they were extracted.

  


  Does anyone has a more reliable way to extract exactly 60 frames from a video with correct timestamps printed on them ?

  


• ffmpeg : Extracted wav from mp4 video does not have equal duration as the original video

  26 juillet 2021, par John Smith

  I have a mp4 video that is 0.92 seconds, and I am trying to extract the audio of the video to a wav format. I have tried several commands (I have provided a list of some of the commands that I have tried), however, the resulting wav does not have the same duration as the original video (the resulting wav often has a duration of 0.96 seconds instead of 0.92 seconds). Ensuring that the video and audio are synchronous is crucial for what I am doing (the videos are typically videos of a person speaking, and it is important that the speech (audio) is in-sync with the mouth movements of the speaker).

  


  I find it odd that, by extracting audio from a video, the duration changes, even despite what is happening under the hood for the conversion (in terms of codecs used, etc).

  


  Some of the commands that I've tried include :

  


  ffmpeg -i <input /> -c copy -map 0:a -sample_rate 16000 <output>&#xA;&#xA;ffmpeg -i <input /> -async  1 -f wav <output>&#xA;&#xA;ffmpeg -i <input /> -vn -acodec copy <output>&#xA;&#xA;ffmpeg -i <input /> -ac 2 -f wav <output>&#xA;</output></output></output></output>

  


  Any insight would be highly appreciated.
Thanks !

  


  Edit Output of the command ffmpeg -ignore_editlist true -i 00026.mp4 output.wav

  


  ffmpeg version 2021-02-28-git-85ab9deb98-full_build-www.gyan.dev Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 10.2.0 (Rev6, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-lib
snappy --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --en
able-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libvidstab --e
nable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libglslang --enable-vulkan --
enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwben
c --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --en
able-libsoxr --enable-chromaprint
  libavutil      56. 66.100 / 56. 66.100
  libavcodec     58.126.100 / 58.126.100
  libavformat    58. 68.100 / 58. 68.100
  libavdevice    58. 12.100 / 58. 12.100
  libavfilter     7.107.100 /  7.107.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '00026.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.37.101
  Duration: 00:00:00.98, start: 0.000000, bitrate: 599 kb/s
  Stream #0:0(und): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 160x160 [SAR 1:1 DAR 1:1], 556 kb/s, 25 fps, 25 tbr, 12800 tbn, 25 tbc (default)
    Metadata:
      handler_name    : VideoHandler
      vendor_id       : [0][0][0][0]
  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 16000 Hz, mono, fltp, 65 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      vendor_id       : [0][0][0][0]
Stream mapping:
  Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'output.wav':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    ISFT            : Lavf58.68.100
  Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      vendor_id       : [0][0][0][0]
      encoder         : Lavc58.126.100 pcm_s16le
size=      32kB time=00:00:00.96 bitrate= 273.7kbits/s speed= 212x
video:0kB audio:32kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.238037%


  


• How can I construct an mp4 with 6 equal audio channels (not 5p1) from an mp4 and a 6 channel wav

  6 août 2021, par ncvp

  OS is Ubuntu Linux.
I start with xxx.webm with picture and synced mono sound.
I construct the 6 channel xxx.wav which has the sound from xxx.webm moved to the 6 channels as appropriate.

  


  % ffmpeg -i xxx.webm -i xxx.wav -map 0:v -map 1:a yyy.mp4

  


  makes the synced yyy.mp4 with 6 audio channels, but the soundtrack is 5p1. Channel 4 is low-pass filtered. This is not what I want.

  


  % ffmpeg -i xxx.webm -i xxx.wav -map 0:v -map 1:a -channel_layout 6.0 yyy.mp4

  


  is an improvement. Channel 4 is not low-pass filtered and it plays perfectly in ffplay and mplayer, but not VLC player.

  


  It turns out there is no channel-layout in yyy.mp4, so there must be something wrong with my -channel_layout 6.0.

  


  Any suggestions, please ?