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• Block artifacts in extracted frame using ffmpeg

  28 août 2020, par 12.yakir

  I'm extracting a single frame from a video with :

  


  ffmpeg -ss 60 -i video.mov -vframes 1 a.jpg


  


  which produces this :

I tried manipulating the pix_fmt and the quality of the q:v 2 to no avail. Note that elsewhere in the video (i.e. not at 60 seconds, but before or after) no such artifacts are visible.

  


  The one thing that helped a little is averaging across multiple frames (adapted from here for 15 frames) :

  


  -vf tmix=frames=15:weights="1 1 1 1 1 1 1 1 1 1 1 1 1 1 1",select='not(mod(n\,15))'


  


  which looks better :

But still, it took a whole 15 frames to get this, and you can still see the artifacts.

  


  Is there a better way to deal with this kind of noise ?

  


  Thank you in advance !

  


  Here are the stats of this video :

  


  ffmpeg version n4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 10.1.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-avisynth --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Downloads/20191130_1h_ICE_2.6m_B9.mov':
  Metadata:
    major_brand     : qt  
    minor_version   : 0
    compatible_brands: qt  
    creation_time   : 2020-03-29T09:24:30.000000Z
  Duration: 00:07:04.83, start: 0.000000, bitrate: 9891 kb/s
    Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 260 kb/s (default)
    Metadata:
      creation_time   : 2020-03-29T09:24:30.000000Z
      handler_name    : Core Media Audio
    Stream #0:1(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 9632 kb/s, 24.38 fps, 25 tbr, 90k tbn, 180k tbc (default)
    Metadata:
      creation_time   : 2020-03-29T09:24:30.000000Z
      handler_name    : Core Media Video
      encoder         : H.264
File 'a.jpg' already exists. Overwrite? [y/N] y
Stream mapping:
  Stream #0:1 -> #0:0 (h264 (native) -> mjpeg (native))
Press [q] to stop, [?] for help
[swscaler @ 0x55a5393a9a40] deprecated pixel format used, make sure you did set range correctly
Output #0, image2, to 'a.jpg':
  Metadata:
    major_brand     : qt  
    minor_version   : 0
    compatible_brands: qt  
    encoder         : Lavf58.45.100
    Stream #0:0(und): Video: mjpeg, yuvj420p(pc), 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc (default)
    Metadata:
      creation_time   : 2020-03-29T09:24:30.000000Z
      handler_name    : Core Media Video
      encoder         : Lavc58.91.100 mjpeg
    Side data:
      cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: N/A
frame=    1 fps=0.0 q=9.3 Lsize=N/A time=00:00:00.04 bitrate=N/A dup=1 drop=1 speed=0.0984x    
video:207kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown


  


• FFMPEG : At least one output file must be specified

  1er décembre 2020, par JeroenWeber

  I am trying to split a video from a URL.... this is my CMD command :

  


  ffmpeg -i 'https://mediasite-ondemand.maastrichtuniversity.nl/MediasiteDeliver/ondemand/MP4Video/90e0ea3d-4087-4a3d-8a78-7acf7906c4cb.mp4?playbackTicket=fb14f2f3baf84412bc6463d4e312ec27&site=mediasite.maastrichtuniversity.nl' -ss 00:00:00 -t 0:25:00 -c copy  SplicedVid.wav


  


  Why does it return :

  


  'site' is not recognized as an internal or external command,
operable program or batch file.

  


  My version :

  


  ffmpeg -version output:
ffmpeg version 4.3.1-2020-11-19-essentials_build-www.gyan.dev Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 10.2.0 (Rev5, 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-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-
libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-libfreetype --enable-libfribidi --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llv
m --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable
-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
libavutil      56. 51.100 / 56. 51.100
libavcodec     58. 91.100 / 58. 91.100
libavformat    58. 45.100 / 58. 45.100
libavdevice    58. 10.100 / 58. 10.100
libavfilter     7. 85.100 /  7. 85.100
libswscale      5.  7.100 /  5.  7.100
libswresample   3.  7.100 /  3.  7.100
libpostproc    55.  7.100 / 55.  7.100


  


• FFmpeg weird codec id for wav_pcm

  1er février 2021, par phongelegob

  So, I want to convert a lot of 32 bit .wav files into 24 bit .wav ones because i want to import them into Omnisphere, a virtual synth. I tried to convert them all with FFmpeg but Omnisphere doesn't support them. I then tested some things and Omnisphere seems to support the audio exported from Edison, which is FL Studio audio editing and recording tool. As I don't want to export them all individually, i wanted to know if there's any way to fix this in FFmpeg or in another audio converter. Here are the test files and what they output in MediaInfo : https://www.mediafire.com/file/l6lojwswsaq9fly/audiotest+wav.zip/file
32 bit original file :

  


  General
Complete name                            : /home/user/Documents/audiotest wav/32bit.wav
Format                                   : Wave
File size                                : 905 KiB
Duration                                 : 2 s 625 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 2 823 kb/s
Writing application                      : Edison
Comment                                  : Recorded on 1/26/2020 in Edison.

Audio
Format                                   : PCM
Format profile                           : Float
Codec ID                                 : 3
Codec ID/Hint                            : IEEE 
Duration                                 : 2 s 625 ms
Bit rate mode                            : Constant
Bit rate                                 : 2 822 kb/s
Channel(s)                               : 2 channels
Sampling rate                            : 44.1 kHz
Bit depth                                : 32 bits
Stream size                              : 904 KiB (100%)


  


  24 bit file exported from Edison :

  


  General
Complete name                            : /home/user/Documents/audiotest wav/edison.wav
Format                                   : Wave
File size                                : 679 KiB
Duration                                 : 2 s 625 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 2 118 kb/s
Writing application                      : Edison
Comment                                  : Recorded on 1/26/2020 in Edison.

Audio
Format                                   : PCM
Format settings                          : Little / Signed
Codec ID                                 : 1
Duration                                 : 2 s 625 ms
Bit rate mode                            : Constant
Bit rate                                 : 2 117 kb/s
Channel(s)                               : 2 channels
Sampling rate                            : 44.1 kHz
Bit depth                                : 24 bits
Stream size                              : 678 KiB (100%)


  


  24 bit file converted with FFmpeg :

  


  General
Complete name                            : /home/user/Documents/audiotest wav/ffmpeg.wav
Format                                   : Wave
File size                                : 679 KiB
Duration                                 : 2 s 625 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 2 117 kb/s
Writing application                      : Lavf58.50.100
Comment                                  : Recorded on 1/26/2020 in Edison.

Audio
Format                                   : PCM
Format settings                          : Little / Signed
Codec ID                                 : 00000001-0000-0010-8000-00AA00389B71
Duration                                 : 2 s 625 ms
Bit rate mode                            : Constant
Bit rate                                 : 2 117 kb/s
Channel(s)                               : 2 channels
Channel layout                           : L R
Sampling rate                            : 44.1 kHz
Bit depth                                : 24 bits
Stream size                              : 678 KiB (100%)