Re: [FFmpeg-user] Confusion about vorbis audio for mkv output

[Paul B Mahol]

On Thu, Feb 18, 2021 at 1:04 AM Ulf Zibis  wrote:

>
> Am 18.02.21 um 00:52 schrieb Paul B Mahol:
> > On Thu, Feb 18, 2021 at 12:50 AM Ulf Zibis  wrote:
> >
> >> Hi,
> >>
> >> Using the following command I get an error:
> >>
> >> The encoder 'vorbis' is experimental but experimental codecs are not
> >> enabled, add '-strict -2' if you want to use it.
> >>
> >> I'm also missing '-strict -2' at
> >> https://ffmpeg.org/ffmpeg-all.html#Codec-Options
> >>
> >> $ ~/Projects/ffmpeg/dev_2/ffmpeg -i INPUT_2.mkv -i Musik.mkv -map 0:v
> -map
> >> 1:a -c:v copy -c:a vorbis INPUT_3.mkv
> >> ffmpeg version N-101141-g8331eb058d Copyright (c) 2000-2021 the FFmpeg
> >> developers
> >> built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
> >> configuration:
> >> libavutil  56. 65.100 / 56. 65.100
> >> libavcodec 58.123.100 / 58.123.100
> >> libavformat58. 67.100 / 58. 67.100
> >> libavdevice58. 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
> >> Input #0, matroska,webm, from 'INPUT_2.mkv':
> >> Metadata:
> >>   SOURCE_TYPE : dvd
> >>   ENCODER : Lavf58.67.100
> >> Duration: 01:31:56.13, start: 0.00, bitrate: 114 kb/s
> >> Chapters:
> >>   Chapter #0:0: start 0.00, end 10.911556
> >>   Chapter #0:1: start 10.911556, end 819.225778
> >>   Chapter #0:2: start 819.225778, end 1839.369956
> >>   Chapter #0:3: start 1839.369956, end 2689.490133
> >>   Chapter #0:4: start 2689.490133, end 3289.574978
> >>   Chapter #0:5: start 3289.574978, end 4584.758089
> >>   Chapter #0:6: start 4584.758089, end 5516.019689
> >> Stream #0:0: Video: mpeg4 (Simple Profile), yuv420p(progressive),
> 16x16
> >> [SAR 16:15 DAR 16:15], 25 fps, 25 tbr, 1k tbn, 25 tbc (default)
> >>   Metadata:
> >> ENCODER : Lavc58.123.100 mpeg4
> >> DURATION: 01:31:56.12000
> >> Stream #0:1(deu): Audio: vorbis, 48000 Hz, stereo, fltp (default)
> >>   Metadata:
> >> LANGUAGE-ISO639_2: deu
> >> LANGUAGE-SIMPLE : Deutsch
> >> LANGUAGE-DESCRIPTION: Deutsch
> >> ENCODER : Lavc58.18.100 libvorbis
> >> DURATION: 01:31:56.13000
> >> [matroska,webm @ 0x5556cb1e5a80] Could not find codec parameters for
> >> stream 0 (Video: h264, none, 426x240): unspecified pixel format
> >> Consider increasing the value for the 'analyzeduration' (0) and
> >> 'probesize' (500) options
> >> Input #1, matroska,webm, from Musik.mkv':
> >> Metadata:
> >>   ENCODER : Lavf58.16.100
> >> Duration: 01:56:18.94, start: -0.007000, bitrate: 108 kb/s
> >> Stream #1:0: Video: h264, none, 426x240, 1k tbr, 1k tbn, 2k tbc
> >> (default)
> >>   Metadata:
> >> HANDLER_NAME: ISO Media file produced by Google Inc. Created
> >> on: 04/18/2020.
> >> DURATION: 00:00:00.0
> >> Stream #1:1(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
> >>   Metadata:
> >> DURATION: 01:56:18.94100
> >> File 'INPUT_3.mkv' already exists. Overwrite? [y/N] y
> >> Stream mapping:
> >> Stream #0:0 -> #0:0 (copy)
> >> Stream #1:1 -> #0:1 (opus (native) -> vorbis (native))
> >> Press [q] to stop, [?] for help
> >> [vorbis @ 0x5556cb205880] The encoder 'vorbis' is experimental but
> >> experimental codecs are not enabled, add '-strict -2' if you want to
> use it.
> >> Conversion failed!
> >>
> >> But when I transcode a mpg video to mkv, then automatically a vorbis
> audio
> >> stream is created, even it is stated "experimental":
> >>
> >> $ ffmpeg -i Kiosk\ Mozartstr.\ -\ Andreas.mpg
> >> Kiosk\ Mozartstr.\ -\ Andreas.mkv
> >> ffmpeg version 4.3.1-static https://johnvansickle.com/ffmpeg/ Copyright
> >> (c) 2000-2020 the FFmpeg developers
> >> built with gcc 8 (Debian 8.3.0-6)
> >> configuration: --enable-gpl --enable-version3 --enable-static
> >> --disable-debug --disable-ffplay --disable-indev=sndio
> >> --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r
> >> --enable-gn

Re: [FFmpeg-user] Confusion about vorbis audio for mkv output

[Paul B Mahol]

On Thu, Feb 18, 2021 at 12:50 AM Ulf Zibis  wrote:

> Hi,
>
> Using the following command I get an error:
>
> The encoder 'vorbis' is experimental but experimental codecs are not
> enabled, add '-strict -2' if you want to use it.
>
> I'm also missing '-strict -2' at
> https://ffmpeg.org/ffmpeg-all.html#Codec-Options
>
> $ ~/Projects/ffmpeg/dev_2/ffmpeg -i INPUT_2.mkv -i Musik.mkv -map 0:v -map
> 1:a -c:v copy -c:a vorbis INPUT_3.mkv
> ffmpeg version N-101141-g8331eb058d Copyright (c) 2000-2021 the FFmpeg
> developers
>built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
>configuration:
>libavutil  56. 65.100 / 56. 65.100
>libavcodec 58.123.100 / 58.123.100
>libavformat58. 67.100 / 58. 67.100
>libavdevice58. 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
> Input #0, matroska,webm, from 'INPUT_2.mkv':
>Metadata:
>  SOURCE_TYPE : dvd
>  ENCODER : Lavf58.67.100
>Duration: 01:31:56.13, start: 0.00, bitrate: 114 kb/s
>Chapters:
>  Chapter #0:0: start 0.00, end 10.911556
>  Chapter #0:1: start 10.911556, end 819.225778
>  Chapter #0:2: start 819.225778, end 1839.369956
>  Chapter #0:3: start 1839.369956, end 2689.490133
>  Chapter #0:4: start 2689.490133, end 3289.574978
>  Chapter #0:5: start 3289.574978, end 4584.758089
>  Chapter #0:6: start 4584.758089, end 5516.019689
>Stream #0:0: Video: mpeg4 (Simple Profile), yuv420p(progressive), 16x16
> [SAR 16:15 DAR 16:15], 25 fps, 25 tbr, 1k tbn, 25 tbc (default)
>  Metadata:
>ENCODER : Lavc58.123.100 mpeg4
>DURATION: 01:31:56.12000
>Stream #0:1(deu): Audio: vorbis, 48000 Hz, stereo, fltp (default)
>  Metadata:
>LANGUAGE-ISO639_2: deu
>LANGUAGE-SIMPLE : Deutsch
>LANGUAGE-DESCRIPTION: Deutsch
>ENCODER : Lavc58.18.100 libvorbis
>DURATION: 01:31:56.13000
> [matroska,webm @ 0x5556cb1e5a80] Could not find codec parameters for
> stream 0 (Video: h264, none, 426x240): unspecified pixel format
> Consider increasing the value for the 'analyzeduration' (0) and
> 'probesize' (500) options
> Input #1, matroska,webm, from Musik.mkv':
>Metadata:
>  ENCODER : Lavf58.16.100
>Duration: 01:56:18.94, start: -0.007000, bitrate: 108 kb/s
>Stream #1:0: Video: h264, none, 426x240, 1k tbr, 1k tbn, 2k tbc
> (default)
>  Metadata:
>HANDLER_NAME: ISO Media file produced by Google Inc. Created
> on: 04/18/2020.
>DURATION: 00:00:00.0
>Stream #1:1(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
>  Metadata:
>DURATION: 01:56:18.94100
> File 'INPUT_3.mkv' already exists. Overwrite? [y/N] y
> Stream mapping:
>Stream #0:0 -> #0:0 (copy)
>Stream #1:1 -> #0:1 (opus (native) -> vorbis (native))
> Press [q] to stop, [?] for help
> [vorbis @ 0x5556cb205880] The encoder 'vorbis' is experimental but
> experimental codecs are not enabled, add '-strict -2' if you want to use it.
> Conversion failed!
>
> But when I transcode a mpg video to mkv, then automatically a vorbis audio
> stream is created, even it is stated "experimental":
>
> $ ~/Projects/ffmpeg/dev_2/ffmpeg -i Kiosk\ Mozartstr.\ -\ Andreas.mpg
> Kiosk\ Mozartstr.\ -\ Andreas.mkv
> ffmpeg version N-101141-g8331eb058d Copyright (c) 2000-2021 the FFmpeg
> developers
>built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
>configuration:
>libavutil  56. 65.100 / 56. 65.100
>libavcodec 58.123.100 / 58.123.100
>libavformat58. 67.100 / 58. 67.100
>libavdevice58. 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
> Input #0, mpeg, from 'Kiosk Mozartstr. - Andreas.mpg':
>Duration: 00:26:00.49, start: 0.529989, bitrate: 1793 kb/s
>Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, bt470bg,
> progressive), 352x288 [SAR 12:11 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50 tbc
>  Side data:
>cpb: bitrate max/min/avg: 9578800/0/0 buffer size: 1835008
> vbv_delay: N/A
>Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 384 kb/s
> File 'Kiosk Mozartstr. - Andreas.mkv' already exists. Overwrite? [y/N] y
> Stream mapping:
>Stream #0:0 -> #0:0 (mpeg2video (native) -> mpeg4 (native))
>Stream #0:1 -> #0:1 (mp2 (native) -> ac3 (native))
> Press [q] to stop, [?] for help
> Output #0, matroska, to 'Kiosk Mozartstr. - Andreas.mkv':
>Metadata:
>  encoder : Lavf58.67.100
>Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p(tv, bt470bg,
> progressive), 352x288 [SAR 12:11 DAR 4:3], q=2-31, 200 kb/s, 25 fps, 1k tbn
>  Metadata:
>encoder : Lavc58.123.100 mpeg4
>  Side data:
>cpb: bitrate max/min/avg: 0/0/20 buffer size: 0 vbv_delay: 

Re: [FFmpeg-user] Naming chapters in Matroska video

[Paul B Mahol]

On Wed, Feb 17, 2021 at 9:36 AM Ulf Zibis  wrote:

>
> Am 17.02.21 um 01:30 schrieb Paul B Mahol:
> >
> > It works here. Note version of libs.
>
> Good morning, here I am again.
>
> Aha, afaics i use the same lib versions, but libpostproc is missing. Is it
> needed? From where I get it?
> Additionally the configuration list is not shown.
>

Have you read carefully -map_metadata documentation?

It is not just single number, but it can have additional arguments.

The only way to reproduce issue could be to upload same but trimmed input
too.


>
> $ ~/Projects/ffmpeg/dev_2/ffmpeg -i INPUT.mkv -i INPUT_test.ffmeta
> -map_metadata 1 -c copy OUTPUT.mkv
> ffmpeg version N-101141-g8331eb058d Copyright (c) 2000-2021 the FFmpeg
> developers
>built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
>configuration:
>libavutil  56. 65.100 / 56. 65.100
>libavcodec 58.123.100 / 58.123.100
>libavformat58. 67.100 / 58. 67.100
>libavdevice58. 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
> Input #0, matroska,webm, from 'INPUT.mkv':
>Metadata:
>  SOURCE_TYPE : dvd
>  ENCODER : Lavf58.12.100
>Duration: 01:31:56.24, start: 0.11, bitrate: 3011 kb/s
>Chapters:
>  Chapter #0:0: start 0.00, end 11.021556
>  Chapter #0:1: start 11.021556, end 819.335778
>  Chapter #0:2: start 819.335778, end 1839.479956
>  Chapter #0:3: start 1839.479956, end 2689.600133
>  Chapter #0:4: start 2689.600133, end 3289.684978
>  Chapter #0:5: start 3289.684978, end 4584.868089
>  Chapter #0:6: start 4584.868089, end 5516.129689
>Stream #0:0: Video: h264 (High), yuv420p(progressive), 720x576 [SAR
> 16:15 DAR 4:3], 25 fps, 25 tbr, 1k tbn, 50 tbc (default)
>  Metadata:
>ENCODER : Lavc58.18.100 libx264
>DURATION: 01:31:56.24000
>Stream #0:1(deu): Audio: vorbis, 48000 Hz, stereo, fltp (default)
>  Metadata:
>LANGUAGE-ISO639_2: deu
>LANGUAGE-SIMPLE : Deutsch
>LANGUAGE-DESCRIPTION: Deutsch
>ENCODER : Lavc58.18.100 libvorbis
>DURATION: 01:31:56.24100
> Input #1, ffmetadata, from 'INPUT_test.ffmeta':
>Metadata:
>  SOURCE_TYPE : dvd
>  encoder : Lavf58.45.100
>  title   : Experiment
>Duration: 01:31:56.02, start: 0.00, bitrate: N/A
>Chapters:
>  Chapter #1:0: start 0.00, end 10.911556
>  Chapter #1:1: start 10.911556, end 819.225778
>Metadata:
>  title   : 1 - Intro
>  Chapter #1:2: start 819.225778, end 1839.369956
>Metadata:
>  title   : 2 - Restaurant
>  Chapter #1:3: start 1839.369956, end 2689.490133
>Metadata:
>  title   : 3 - At home
>  Chapter #1:4: start 2689.490133, end 3289.574978
>  Chapter #1:5: start 3289.574978, end 4584.758089
>  Chapter #1:6: start 4584.758089, end 5516.019689
> File 'OUTPUT.mkv' already exists. Overwrite? [y/N] y
> Output #0, matroska, to 'OUTPUT.mkv':
>Metadata:
>  SOURCE_TYPE : dvd
>  title   : Experiment
>  encoder : Lavf58.67.100
>Chapters:
>  Chapter #0:0: start 0.00, end 10.911556
>  Chapter #0:1: start 10.911556, end 819.225778
>  Chapter #0:2: start 819.225778, end 1839.369956
>  Chapter #0:3: start 1839.369956, end 2689.490133
>  Chapter #0:4: start 2689.490133, end 3289.574978
>  Chapter #0:5: start 3289.574978, end 4584.758089
>  Chapter #0:6: start 4584.758089, end 5516.019689
>Stream #0:0: Video: h264 (High) (H264 / 0x34363248),
> yuv420p(progressive), 720x576 [SAR 16:15 DAR 4:3], q=2-31, 25 fps, 25 tbr,
> 1k tbn, 1k tbc (default)
>  Metadata:
>ENCODER : Lavc58.18.100 libx264
>DURATION: 01:31:56.24000
>Stream #0:1(deu): Audio: vorbis (oV[0][0] / 0x566F), 48000 Hz, stereo,
> fltp (default)
>  Metadata:
>LANGUAGE-ISO639_2: deu
>LANGUAGE-SIMPLE : Deutsch
>LANGUAGE-DESCRIPTION: Deutsch
>ENCODER : Lavc58.18.100 libvorbis
>DURATION: 01:31:56.24100
> Stream mapping:
>Stream #0:0 -> #0:0 (copy)
>Stream #0:1 -> #0:1 (copy)
> Press [q] to stop, [?] for help
> frame=137903 fps=871 q=-1.0 Lsize= 2027570kB time=01:31:56.12
> bitrate=3011.1kbits/s speed=34.8x
> video:1952414kB audio:71785kB subtitle:0kB other streams:0kB global
> headers:4kB muxing overhead: 0.166526%
>
> $ ~/Projects/ffmpeg/dev_2/ffprobe OUTPUT.mkv
> ffprobe version N-101141-

Re: [FFmpeg-user] Naming chapters in Matroska video

[Paul B Mahol]

On Wed, Feb 17, 2021 at 1:26 AM Ulf Zibis  wrote:

>
> Am 17.02.21 um 01:20 schrieb Paul B Mahol:
> > ffprobe version N-101155-g96b5919034 Copyright (c) 2007-2021 the FFmpeg
> > developers
> >built with Ubuntu clang version
> > 11.1.0-++20210203115409+1fdec59bffc1-1~exp1~20210203230038.161
> >configuration: --cc='ccache clang-11' --as='ccache clang-11'
> > --disable-indevs --disable-outdevs --disable-autodetect --enable-sdl2
> > --enable-gpl --enable-libpulse --enable-indev=lavfi --enable-indev=alsa
> > --enable-indev=puls
> > e --enable-libfdk-aac --enable-nonfree --enable-zlib --enable-openssl
> > --enable-outdev=alsa --disable-static --enable-shared --enable-ladspa
> > --enable-lv2 --enable-frei0r --enable-liblensfun --enable-version3
> > --enable-libdav1d
> >   --enable-libbs2b --enable-libtwolame --enable-libmp3lame
> > --enable-librubberband --enable-libmysofa --enable-libx264
> > --enable-fontconfig --enable-libfreetype --enable-libass --enable-libzimg
> > --enable-librist --disable-stripp
> > ing --samples=../fate-suite
> >libavutil  56. 65.100 / 56. 65.100
> >libavcodec 58.123.100 / 58.123.100
> >libavformat58. 67.100 / 58. 67.100
> >libavdevice58. 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
> >libpostproc55.  8.100 / 55.  8.100
> > Input #0, matroska,webm, from 'x.mkv':
> >Metadata:
> >  title   : Experiment
> >  SOURCE_TYPE : dvd
> >  ENCODER : Lavf58.67.100
> >Duration: 00:00:11.00, start: 0.00, bitrate: 10220 kb/s
> >Chapters:
> >  Chapter #0:0: start 0.00, end 10.911556
> >  Chapter #0:1: start 10.911556, end 819.225778
> >Metadata:
> >  title   : 1 - Intro
> >  Chapter #0:2: start 819.225778, end 1839.369956
> >Metadata:
> >  title   : 2 - Restaurant
> >  Chapter #0:3: start 1839.369956, end 2689.490133
> >Metadata:
> >  title   : 3 - At home
> >  Chapter #0:4: start 2689.490133, end 3289.574978
> >  Chapter #0:5: start 3289.574978, end 4584.758089
> >  Chapter #0:6: start 4584.758089, end 5516.019689
> >Stream #0:0: Video: mjpeg (Baseline), yuvj422p(pc,
> > bt470bg/unknown/unknown), 1280x720, 24 fps, 24 tbr, 1k tbn, 1k tbc
> (default)
> >  Metadata:
> >DURATION: 00:00:11.0
> >Stream #0:1: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s
> > (default)
> >  Metadata:
> >DURATION: 00:00:11.0
>
> What do you want to say me with this?
>

It works here. Note version of libs.


>
> -Ulf
>
> ___
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Re: [FFmpeg-user] Naming chapters in Matroska video

[Paul B Mahol]

ffprobe version N-101155-g96b5919034 Copyright (c) 2007-2021 the FFmpeg
developers
  built with Ubuntu clang version
11.1.0-++20210203115409+1fdec59bffc1-1~exp1~20210203230038.161
  configuration: --cc='ccache clang-11' --as='ccache clang-11'
--disable-indevs --disable-outdevs --disable-autodetect --enable-sdl2
--enable-gpl --enable-libpulse --enable-indev=lavfi --enable-indev=alsa
--enable-indev=puls
e --enable-libfdk-aac --enable-nonfree --enable-zlib --enable-openssl
--enable-outdev=alsa --disable-static --enable-shared --enable-ladspa
--enable-lv2 --enable-frei0r --enable-liblensfun --enable-version3
--enable-libdav1d
 --enable-libbs2b --enable-libtwolame --enable-libmp3lame
--enable-librubberband --enable-libmysofa --enable-libx264
--enable-fontconfig --enable-libfreetype --enable-libass --enable-libzimg
--enable-librist --disable-stripp
ing --samples=../fate-suite
  libavutil  56. 65.100 / 56. 65.100
  libavcodec 58.123.100 / 58.123.100
  libavformat58. 67.100 / 58. 67.100
  libavdevice58. 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
  libpostproc55.  8.100 / 55.  8.100
Input #0, matroska,webm, from 'x.mkv':
  Metadata:
title   : Experiment
SOURCE_TYPE : dvd
ENCODER : Lavf58.67.100
  Duration: 00:00:11.00, start: 0.00, bitrate: 10220 kb/s
  Chapters:
Chapter #0:0: start 0.00, end 10.911556
Chapter #0:1: start 10.911556, end 819.225778
  Metadata:
title   : 1 - Intro
Chapter #0:2: start 819.225778, end 1839.369956
  Metadata:
title   : 2 - Restaurant
Chapter #0:3: start 1839.369956, end 2689.490133
  Metadata:
title   : 3 - At home
Chapter #0:4: start 2689.490133, end 3289.574978
Chapter #0:5: start 3289.574978, end 4584.758089
Chapter #0:6: start 4584.758089, end 5516.019689
  Stream #0:0: Video: mjpeg (Baseline), yuvj422p(pc,
bt470bg/unknown/unknown), 1280x720, 24 fps, 24 tbr, 1k tbn, 1k tbc (default)
Metadata:
  DURATION: 00:00:11.0
  Stream #0:1: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s
(default)
Metadata:
  DURATION: 00:00:11.0

On Wed, Feb 17, 2021 at 1:13 AM Ulf Zibis  wrote:

>
> Am 17.02.21 um 01:02 schrieb Paul B Mahol:
> > On Wed, Feb 17, 2021 at 12:58 AM Ulf Zibis  wrote:
> >
> >>
> >> I still get the same result with the latest nightly. :-(
> >>
> >> Please share your .ffmeta file.
>
> See attachment!
>
> -Ulf
>
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
> To unsubscribe, visit link above, or email
> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
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Re: [FFmpeg-user] Naming chapters in Matroska video

[Paul B Mahol]

On Wed, Feb 17, 2021 at 12:58 AM Ulf Zibis  wrote:

>
> Am 16.02.21 um 23:49 schrieb Paul B Mahol:
> > I do not use release versions, so try latest nightly version instead.
>
> I still get the same result with the latest nightly. :-(
>
> Please share your .ffmeta file.

> -Ulf
>
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Re: [FFmpeg-user] Naming chapters in Matroska video

[Paul B Mahol]

On Wed, Feb 17, 2021 at 12:27 AM Ulf Zibis  wrote:

>
> Am 16.02.21 um 23:20 schrieb Paul B Mahol:
> > I used .ffmeta as extension for metadata and it worked fine.
> If the extension is important, why isn't this documented here? :
>

Because it is probed and if it is probed as ffmeta you have nothing to
worry.
.ffmeta is just correct extension unlike .meta


>
> https://ffmpeg.org/ffmpeg-all.html#Metadata-1
>
> -Ulf
>
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Re: [FFmpeg-user] converting raw stills to "raw" video

[Paul B Mahol]

On Wed, Feb 17, 2021 at 12:01 AM Michael Koch 
wrote:

> Am 16.02.2021 um 23:52 schrieb Paul B Mahol:
> > On Tue, Feb 16, 2021 at 11:47 PM Michael Koch <
> astroelectro...@t-online.de>
> > wrote:
> >
> >> Am 16.02.2021 um 21:34 schrieb Paul B Mahol:
> >>> On Tue, Feb 16, 2021 at 9:28 PM Michael Koch <
> >> astroelectro...@t-online.de>
> >>> wrote:
> >>>
> >>>> Am 16.02.2021 um 11:52 schrieb shawny:
> >>>>> I am trying to find out if it is possible using ffmpeg to convert raw
> >>>> stills
> >>>>> into "raw" or log video in order to be gradable as video.  This is to
> >>>>> incorporate timelapse photography into my video projects.  I know
> this
> >>>> can
> >>>>> be done using Adobe After Effects but I am trying to find a post path
> >>>> that
> >>>>> uses Linux and open source software.
> >>>>> Has anyone done this or is anyone able to tell me one way or another
> >>>> whether
> >>>>> it is currently possible with ffmpeg?
> >>>> A possible path with open source software is to use "Gimp" with
> >>>> "Darktable" plugin to convert the raw images to 16-bit PNG. Then you
> can
> >>>> use FFmpeg to combine these images to a video.
> >>>>
> >>> Stop spreading lies.
> >> I don't know what's wrong with my answer. Please explain.
> >>
> > Decoder works fine, it is just swscale does not support right colorspace
> > conversion.
> >
> > Use zscale filter instead. Make sure that you set correct parameters,
> > parameters depends
> > on input DNG file.
>
> The question wasn't about DNG files. It was about raw files in general.
> My answer was correct.
>

No, it is misleading and incorrect. Also RAW is very broad subject.
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Re: [FFmpeg-user] converting raw stills to "raw" video

[Paul B Mahol]

On Tue, Feb 16, 2021 at 11:47 PM Michael Koch 
wrote:

> Am 16.02.2021 um 21:34 schrieb Paul B Mahol:
> > On Tue, Feb 16, 2021 at 9:28 PM Michael Koch <
> astroelectro...@t-online.de>
> > wrote:
> >
> >> Am 16.02.2021 um 11:52 schrieb shawny:
> >>> I am trying to find out if it is possible using ffmpeg to convert raw
> >> stills
> >>> into "raw" or log video in order to be gradable as video.  This is to
> >>> incorporate timelapse photography into my video projects.  I know this
> >> can
> >>> be done using Adobe After Effects but I am trying to find a post path
> >> that
> >>> uses Linux and open source software.
> >>> Has anyone done this or is anyone able to tell me one way or another
> >> whether
> >>> it is currently possible with ffmpeg?
> >> A possible path with open source software is to use "Gimp" with
> >> "Darktable" plugin to convert the raw images to 16-bit PNG. Then you can
> >> use FFmpeg to combine these images to a video.
> >>
> > Stop spreading lies.
>
> I don't know what's wrong with my answer. Please explain.
>

Decoder works fine, it is just swscale does not support right colorspace
conversion.

Use zscale filter instead. Make sure that you set correct parameters,
parameters depends
on input DNG file.


>
> Michael
>
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Re: [FFmpeg-user] Naming chapters in Matroska video

[Paul B Mahol]

I do not use release versions, so try latest nightly version instead.
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Re: [FFmpeg-user] Naming chapters in Matroska video

[Paul B Mahol]

On Tue, Feb 16, 2021 at 10:51 PM Ulf Zibis  wrote:

> Hi,
>
> This is the meta data from my original video:
>
> ;FFMETADATA1
> SOURCE_TYPE=dvd
> encoder=Lavf58.45.100
> [CHAPTER]
> TIMEBASE=1/10
> START=0
> END=1091156
> [CHAPTER]
> TIMEBASE=1/10
> START=1091156
> END=81922578
> [CHAPTER]
> TIMEBASE=1/10
> START=81922578
> END=183936996
> [CHAPTER]
> TIMEBASE=1/10
> START=183936996
> END=268949013
> [CHAPTER]
> TIMEBASE=1/10
> START=268949013
> END=328957498
> [CHAPTER]
> TIMEBASE=1/10
> START=328957498
> END=458475809
> [CHAPTER]
> TIMEBASE=1/10
> START=458475809
> END=551601969
>
> I've edited the file to (added chapter names):
>
> ;FFMETADATA1
> SOURCE_TYPE=dvd
> encoder=Lavf58.45.100
> [CHAPTER]
> TIMEBASE=1/10
> START=0
> END=1091156
> [CHAPTER]
> TIMEBASE=1/10
> START=1091156
> END=81922578
> title=1 - Intro
> [CHAPTER]
> TIMEBASE=1/10
> START=81922578
> END=183936996
> title=2 - Restaurant
> [CHAPTER]
> TIMEBASE=1/10
> START=183936996
> END=268949013
> title=3 - At home
> [.]
>
> After importing it to the video, with
>
> ffmpeg -i INPUT.mkv -i INPUT.meta -map_metadata 1 -c copy -movflags
> +faststart OUTPUT.mkv
>

movflags +faststart for mkv does nothing.


>
> , ... I still don't see the named chapters.
>
> How to correctly add chapter names?
>

I used .ffmeta as extension for metadata and it worked fine.


>
> Here the ffprobe output:
>
> $ ffprobe OUTPUT.mkv
> ffprobe version 4.3.1-static https://johnvansickle.com/ffmpeg/ Copyright
> (c) 2007-2020 the FFmpeg developers
>built with gcc 8 (Debian 8.3.0-6)
>configuration: --enable-gpl --enable-version3 --enable-static
> --disable-debug --disable-ffplay --disable-indev=sndio
> --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r
> --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom
> --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype
> --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libopenjpeg --enable-librubberband --enable-libsoxr
> --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus
> --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc
> --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi
> --enable-libzimg
>libavutil  56. 51.100 / 56. 51.100
>libavcodec 58. 91.100 / 58. 91.100
>libavformat58. 45.100 / 58. 45.100
>libavdevice58. 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
>libpostproc55.  7.100 / 55.  7.100
> Input #0, matroska,webm, from 'OUTPUT.mkv':
>Metadata:
>  SOURCE_TYPE : dvd
>  ENCODER : Lavf58.45.100
>Duration: 01:31:56.13, start: 0.00, bitrate: 3011 kb/s
>  Chapter #0:0: start 0.00, end 10.911556
>  Chapter #0:1: start 10.911556, end 819.225778
>  Chapter #0:2: start 819.225778, end 1839.369956
>  Chapter #0:3: start 1839.369956, end 2689.490133
>  Chapter #0:4: start 2689.490133, end 3289.574978
>  Chapter #0:5: start 3289.574978, end 4584.758089
>  Chapter #0:6: start 4584.758089, end 5516.019689
>  Stream #0:0: Video: h264 (High), yuv420p(progressive), 720x576 [SAR
> 16:15 DAR 4:3], 25 fps, 25 tbr, 1k tbn, 50 tbc (default)
>  Metadata:
>ENCODER : Lavc58.18.100 libx264
>DURATION: 01:31:56.13000
>  Stream #0:1(deu): Audio: vorbis, 48000 Hz, stereo, fltp (default)
>  Metadata:
>LANGUAGE-ISO639_2: deu
>LANGUAGE-SIMPLE : Deutsch
>LANGUAGE-DESCRIPTION: Deutsch
>ENCODER : Lavc58.18.100 libvorbis
>DURATION: 01:31:56.13000
>
> -Ulf
>
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Re: [FFmpeg-user] converting raw stills to "raw" video

[Paul B Mahol]

On Tue, Feb 16, 2021 at 9:28 PM Michael Koch 
wrote:

> Am 16.02.2021 um 11:52 schrieb shawny:
> > I am trying to find out if it is possible using ffmpeg to convert raw
> stills
> > into "raw" or log video in order to be gradable as video.  This is to
> > incorporate timelapse photography into my video projects.  I know this
> can
> > be done using Adobe After Effects but I am trying to find a post path
> that
> > uses Linux and open source software.
> > Has anyone done this or is anyone able to tell me one way or another
> whether
> > it is currently possible with ffmpeg?
>
> A possible path with open source software is to use "Gimp" with
> "Darktable" plugin to convert the raw images to 16-bit PNG. Then you can
> use FFmpeg to combine these images to a video.
>

Stop spreading lies.


>
> Michael
>
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Re: [FFmpeg-user] Same output bit depth and sampling frequency as input?

[Paul B Mahol]

On Mon, Feb 15, 2021 at 5:03 PM Marco Mircoli  wrote:

> I think it is not a bug.
> It is normal that dynaudnorm and loudnorm gives differents results, they
> are 2 different instruments and they have different parameters and actions.
>

I never claimed different. Using both filters at same time does not make
sense.


> Thanks.
> S.
>
>
>
> Il giorno dom 14 feb 2021 alle ore 12:08 Paul B Mahol 
> ha
> scritto:
>
> > On Sun, Feb 14, 2021 at 12:01 PM Marco Mircoli 
> > wrote:
> >
> > > Yes, the result Is not leveled to the target loudness value
> > >
> >
> > Than that issue should be reported to our issue tracker, as loudnorm bug.
> >
> >
> > >
> > > Il Ven 12 Feb 2021, 19:26 Paul B Mahol  ha scritto:
> > >
> > > > On Fri, Feb 12, 2021 at 7:17 PM Marco Mircoli 
> > > > wrote:
> > > >
> > > > > Il giorno ven 12 feb 2021 alle ore 10:53 Paul B Mahol <
> > > one...@gmail.com>
> > > > > ha
> > > > > scritto:
> > > > >
> > > > > > On Fri, Feb 12, 2021 at 12:48 AM Marco Mircoli <
> > nuvolab...@gmail.com
> > > >
> > > > > > wrote:
> > > > > >
> > > > > > > Hello everybody,
> > > > > > >  wondering how to set the same output audio specs (sample rate
> > > anche
> > > > > bit
> > > > > > > depth) as input.
> > > > > > >
> > > > > > > I tried like that
> > > > > > >
> > > > > > > ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
> > > > > > > 5:TP=-1.5:LRA=7 -sample_fmt s16 -ar 44100 FILE_TO
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > if I have a 48KHZ input, the output is 44100, so doesn't work
> for
> > > my
> > > > > > needs
> > > > > > >
> > > > > > > Tried without
> > > > > > >
> > > > > > >
> > > > > > > ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
> > > > > > > 5:TP=-1.5:LRA=7  FILE_TO
> > > > > > >
> > > > > > > The result is: same bit depth of input file but 192Khz sample
> > rate.
> > > > > (file
> > > > > > > input sr is 48khz)
> > > > > > >
> > > > > > > Anybody can give me a suggest?
> > > > > > >
> > > > > >
> > > > > > Use shell scripting.
> > > > > >
> > > > > > Also using those 2 filters at same time is entirely invalid and
> > > > > pointless.
> > > > > >
> > > > > >
> > > > > >
> > > > > Unfortunately, don't know ho to use shell. I'm a newbe. Any
> resource
> > > that
> > > > > can help me to start with shell scriping?
> > > > >
> > > > > ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.5:TP=-1.5:LRA=7
> > > > FILE_TO
> > > > >
> > > > > Cannot understand why this Is invalid.
> > > > > 1st filter (dynaudnorm), level the audio changing the volume every
> > > frame
> > > > > window
> > > > > 2nd filter (loudnorm) set an offset to the gain to let the loudness
> > go
> > > to
> > > > > 16.5lkfs
> > > > >
> > > > > I've miss something?
> > > > >
> > > >
> > > > Is there any difference when you remove dynaudnorm?
> > > >
> > > >
> > > > >
> > > > > Thanks.
> > > > > S:
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > >
> > > > > > > Thanks,
> > > > > > > S.
> > > > > > > ___
> > > > > > > ffmpeg-user mailing list
> > > > > > > ffmpeg-user@ffmpeg.org
> > > > > > > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > > > > > >
> > > > > > > To unsubscribe, visit link above, or email
> > > > > > > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
&g

Re: [FFmpeg-user] Filter documentation -- PTSs

[Paul B Mahol]

On Mon, Feb 15, 2021 at 2:39 AM Mark Filipak (ffmpeg) 
wrote:

> On 02/14/2021 08:28 PM, Paul B Mahol wrote:
> > On Mon, Feb 15, 2021 at 2:23 AM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 02/14/2021 08:13 PM, Paul B Mahol wrote:
> >>> On Mon, Feb 15, 2021 at 1:46 AM Mark Filipak (ffmpeg) <
> >> markfili...@bog.us>
> >>> wrote:
> >>>
> >>>> On 02/14/2021 07:34 PM, Paul B Mahol wrote:
> >>>>> On Mon, Feb 15, 2021 at 12:50 AM Mark Filipak (ffmpeg) <
> >>>> markfili...@bog.us>
> >>>>> wrote:
> >>>>>
> >>>>>> Disclosure: Everything that follows may be wrong.
> >>>>>>
> >>>>>> I reckon that, except for a few filters, the important metric is
> PTS,
> >>>> not
> >>>>>> the frame #s assigned at a
> >>>>>> filter's input node nor its wall-clock arrival time at a filter's
> >> input
> >>>>>> node. For example,
> >>>>>> 'interleave' performs frame interleaves based on the PTSs of
> incoming
> >>>>>> frames -- at least, that's
> >>>>>> what I think.
> >>>>>>
> >>>>>> Am I correct?
> >>>>>>
> >>>>>> If so, wouldn't it be helpful if filter documentation included how
> it
> >>>>>> generates PTSs?
> >>>>>>
> >>>>>
> >>>>> PTS are not generated.
> >>>>
> >>>> ffmpeg -i SOURCE -vf telecine TARGET
> >>>>
> >>>> What are the PTSs of the first 5 frames at the output of the filter?
> >>>>
> >>>>
> >>> Interpolated by very complex formula.
> >>
> >> So, the PTSs are generated.
> >>
> >> What is the formula? Certainly it can't be anywhere nearly as complex as
> >> the differential equations
> >> and boundary conditions that control air flow through a duct of
> arbitrary
> >> shape or the air flow
> >> across an aircraft wing of arbitrary profile. I guess if I can handle
> >> those, I'll be able to handle
> >> the generated PTSs of a periodic series of frames, eh?
> >>
> >
> > You need TB aka timebase too.
> >
> > TB is rational.
> > PTS is 64bit integer.
>
> Thank you, Paul. I promise I can handle it. What is the formula?
>
>
See source code of telecine filter. It is doing all calculations avoiding
floats.
But formula in floats are pretty trivial and straightforward to write.

Note that when you change pts in wrong way like you do, you need to change
timebase too
or you will get many problems out of nowhere.



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Re: [FFmpeg-user] Filter documentation -- PTSs

[Paul B Mahol]

On Mon, Feb 15, 2021 at 2:23 AM Mark Filipak (ffmpeg) 
wrote:

> On 02/14/2021 08:13 PM, Paul B Mahol wrote:
> > On Mon, Feb 15, 2021 at 1:46 AM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 02/14/2021 07:34 PM, Paul B Mahol wrote:
> >>> On Mon, Feb 15, 2021 at 12:50 AM Mark Filipak (ffmpeg) <
> >> markfili...@bog.us>
> >>> wrote:
> >>>
> >>>> Disclosure: Everything that follows may be wrong.
> >>>>
> >>>> I reckon that, except for a few filters, the important metric is PTS,
> >> not
> >>>> the frame #s assigned at a
> >>>> filter's input node nor its wall-clock arrival time at a filter's
> input
> >>>> node. For example,
> >>>> 'interleave' performs frame interleaves based on the PTSs of incoming
> >>>> frames -- at least, that's
> >>>> what I think.
> >>>>
> >>>> Am I correct?
> >>>>
> >>>> If so, wouldn't it be helpful if filter documentation included how it
> >>>> generates PTSs?
> >>>>
> >>>
> >>> PTS are not generated.
> >>
> >> ffmpeg -i SOURCE -vf telecine TARGET
> >>
> >> What are the PTSs of the first 5 frames at the output of the filter?
> >>
> >>
> > Interpolated by very complex formula.
>
> So, the PTSs are generated.
>
> What is the formula? Certainly it can't be anywhere nearly as complex as
> the differential equations
> and boundary conditions that control air flow through a duct of arbitrary
> shape or the air flow
> across an aircraft wing of arbitrary profile. I guess if I can handle
> those, I'll be able to handle
> the generated PTSs of a periodic series of frames, eh?
>

You need TB aka timebase too.

TB is rational.
PTS is 64bit integer.


>
> --
> Any journey, no matter how long, is just a series of small steps.
> "Government is the problem!" -- 1982 and onward.
> "___ is the enemy of the people!" -- 2016 and onward.
> "You have to fight like hell or you're not going to have a country!" --
> Jan 6, 2021.
> It isn't the distance that's important, it's the direction.
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Re: [FFmpeg-user] Filter documentation -- PTSs

[Paul B Mahol]

On Mon, Feb 15, 2021 at 2:19 AM Phil Rhodes via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

>  How does all this interact with files from cameras with very stable
> timing, such as broadcast and cinema cameras?
> Is there not a risk of long-term rounding errors?
>

PTS and TB (timebase) are used together.


> P
> On Monday, 15 February 2021, 01:13:37 GMT, Paul B Mahol <
> one...@gmail.com> wrote:
>
>  On Mon, Feb 15, 2021 at 1:46 AM Mark Filipak (ffmpeg)  >
> wrote:
>
> > On 02/14/2021 07:34 PM, Paul B Mahol wrote:
> > > On Mon, Feb 15, 2021 at 12:50 AM Mark Filipak (ffmpeg) <
> > markfili...@bog.us>
> > > wrote:
> > >
> > >> Disclosure: Everything that follows may be wrong.
> > >>
> > >> I reckon that, except for a few filters, the important metric is PTS,
> > not
> > >> the frame #s assigned at a
> > >> filter's input node nor its wall-clock arrival time at a filter's
> input
> > >> node. For example,
> > >> 'interleave' performs frame interleaves based on the PTSs of incoming
> > >> frames -- at least, that's
> > >> what I think.
> > >>
> > >> Am I correct?
> > >>
> > >> If so, wouldn't it be helpful if filter documentation included how it
> > >> generates PTSs?
> > >>
> > >
> > > PTS are not generated.
> >
> > ffmpeg -i SOURCE -vf telecine TARGET
> >
> > What are the PTSs of the first 5 frames at the output of the filter?
> >
> >
> Interpolated by very complex formula.
>
>
> > --
> > Any journey, no matter how long, is just a series of small steps.
> > "Government is the problem!" -- 1982 and onward.
> > "___ is the enemy of the people!" -- 2016 and onward.
> > "You have to fight like hell or you're not going to have a country!" --
> > Jan 6, 2021.
> > It isn't the distance that's important, it's the direction.
> > ___
> > ffmpeg-user mailing list
> > ffmpeg-user@ffmpeg.org
> > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
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Re: [FFmpeg-user] Filter documentation -- PTSs

[Paul B Mahol]

On Mon, Feb 15, 2021 at 1:46 AM Mark Filipak (ffmpeg) 
wrote:

> On 02/14/2021 07:34 PM, Paul B Mahol wrote:
> > On Mon, Feb 15, 2021 at 12:50 AM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> Disclosure: Everything that follows may be wrong.
> >>
> >> I reckon that, except for a few filters, the important metric is PTS,
> not
> >> the frame #s assigned at a
> >> filter's input node nor its wall-clock arrival time at a filter's input
> >> node. For example,
> >> 'interleave' performs frame interleaves based on the PTSs of incoming
> >> frames -- at least, that's
> >> what I think.
> >>
> >> Am I correct?
> >>
> >> If so, wouldn't it be helpful if filter documentation included how it
> >> generates PTSs?
> >>
> >
> > PTS are not generated.
>
> ffmpeg -i SOURCE -vf telecine TARGET
>
> What are the PTSs of the first 5 frames at the output of the filter?
>
>
Interpolated by very complex formula.


> --
> Any journey, no matter how long, is just a series of small steps.
> "Government is the problem!" -- 1982 and onward.
> "___ is the enemy of the people!" -- 2016 and onward.
> "You have to fight like hell or you're not going to have a country!" --
> Jan 6, 2021.
> It isn't the distance that's important, it's the direction.
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
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Re: [FFmpeg-user] Filter documentation -- PTSs

[Paul B Mahol]

On Mon, Feb 15, 2021 at 12:50 AM Mark Filipak (ffmpeg) 
wrote:

> Disclosure: Everything that follows may be wrong.
>
> I reckon that, except for a few filters, the important metric is PTS, not
> the frame #s assigned at a
> filter's input node nor its wall-clock arrival time at a filter's input
> node. For example,
> 'interleave' performs frame interleaves based on the PTSs of incoming
> frames -- at least, that's
> what I think.
>
> Am I correct?
>
> If so, wouldn't it be helpful if filter documentation included how it
> generates PTSs?
>

PTS are not generated.


>
> Filter documentation should include:
> 1, More than just the names of directives (i.e. "options") -- some do,
> some don't,
> 2, Input specifications that include
> 2.1, The minimum number of output frames,
> 2.2, The number of output frames per input frame, and
> 2.3, Indication of the frames that are dropped (if any), for example:
> "first frame is dropped",
> "last 2 frames are dropped", etc., and
> 3, The mathematical formula employed to calculate output PTSs.
>
> I urge responses from all who are critical -- You know who you are!
> Perhaps someday we will be
> allowed to contribute. There are several of us who are critical. If
> allowed, we could resolve all
> issues for all filters in just a few days.
>
> --
> Any journey, no matter how long, is just a series of small steps.
> "Government is the problem!" -- 1982 and onward.
> "___ is the enemy of the people!" -- 2016 and onward.
> "You have to fight like hell or you're not going to have a country!" --
> Jan 6, 2021.
> It isn't the distance that's important, it's the direction.
> ___
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> ffmpeg-user@ffmpeg.org
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Re: [FFmpeg-user] Same output bit depth and sampling frequency as input?

[Paul B Mahol]

On Sun, Feb 14, 2021 at 12:01 PM Marco Mircoli  wrote:

> Yes, the result Is not leveled to the target loudness value
>

Than that issue should be reported to our issue tracker, as loudnorm bug.


>
> Il Ven 12 Feb 2021, 19:26 Paul B Mahol  ha scritto:
>
> > On Fri, Feb 12, 2021 at 7:17 PM Marco Mircoli 
> > wrote:
> >
> > > Il giorno ven 12 feb 2021 alle ore 10:53 Paul B Mahol <
> one...@gmail.com>
> > > ha
> > > scritto:
> > >
> > > > On Fri, Feb 12, 2021 at 12:48 AM Marco Mircoli  >
> > > > wrote:
> > > >
> > > > > Hello everybody,
> > > > >  wondering how to set the same output audio specs (sample rate
> anche
> > > bit
> > > > > depth) as input.
> > > > >
> > > > > I tried like that
> > > > >
> > > > > ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
> > > > > 5:TP=-1.5:LRA=7 -sample_fmt s16 -ar 44100 FILE_TO
> > > > >
> > > > >
> > > > >
> > > > > if I have a 48KHZ input, the output is 44100, so doesn't work for
> my
> > > > needs
> > > > >
> > > > > Tried without
> > > > >
> > > > >
> > > > > ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
> > > > > 5:TP=-1.5:LRA=7  FILE_TO
> > > > >
> > > > > The result is: same bit depth of input file but 192Khz sample rate.
> > > (file
> > > > > input sr is 48khz)
> > > > >
> > > > > Anybody can give me a suggest?
> > > > >
> > > >
> > > > Use shell scripting.
> > > >
> > > > Also using those 2 filters at same time is entirely invalid and
> > > pointless.
> > > >
> > > >
> > > >
> > > Unfortunately, don't know ho to use shell. I'm a newbe. Any resource
> that
> > > can help me to start with shell scriping?
> > >
> > > ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.5:TP=-1.5:LRA=7
> > FILE_TO
> > >
> > > Cannot understand why this Is invalid.
> > > 1st filter (dynaudnorm), level the audio changing the volume every
> frame
> > > window
> > > 2nd filter (loudnorm) set an offset to the gain to let the loudness go
> to
> > > 16.5lkfs
> > >
> > > I've miss something?
> > >
> >
> > Is there any difference when you remove dynaudnorm?
> >
> >
> > >
> > > Thanks.
> > > S:
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > >
> > > > > Thanks,
> > > > > S.
> > > > > ___
> > > > > ffmpeg-user mailing list
> > > > > ffmpeg-user@ffmpeg.org
> > > > > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > > > >
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Re: [FFmpeg-user] Same output bit depth and sampling frequency as input?

[Paul B Mahol]

On Fri, Feb 12, 2021 at 7:17 PM Marco Mircoli  wrote:

> Il giorno ven 12 feb 2021 alle ore 10:53 Paul B Mahol 
> ha
> scritto:
>
> > On Fri, Feb 12, 2021 at 12:48 AM Marco Mircoli 
> > wrote:
> >
> > > Hello everybody,
> > >  wondering how to set the same output audio specs (sample rate anche
> bit
> > > depth) as input.
> > >
> > > I tried like that
> > >
> > > ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
> > > 5:TP=-1.5:LRA=7 -sample_fmt s16 -ar 44100 FILE_TO
> > >
> > >
> > >
> > > if I have a 48KHZ input, the output is 44100, so doesn't work for my
> > needs
> > >
> > > Tried without
> > >
> > >
> > > ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
> > > 5:TP=-1.5:LRA=7  FILE_TO
> > >
> > > The result is: same bit depth of input file but 192Khz sample rate.
> (file
> > > input sr is 48khz)
> > >
> > > Anybody can give me a suggest?
> > >
> >
> > Use shell scripting.
> >
> > Also using those 2 filters at same time is entirely invalid and
> pointless.
> >
> >
> >
> Unfortunately, don't know ho to use shell. I'm a newbe. Any resource that
> can help me to start with shell scriping?
>
> ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.5:TP=-1.5:LRA=7  FILE_TO
>
> Cannot understand why this Is invalid.
> 1st filter (dynaudnorm), level the audio changing the volume every frame
> window
> 2nd filter (loudnorm) set an offset to the gain to let the loudness go to
> 16.5lkfs
>
> I've miss something?
>

Is there any difference when you remove dynaudnorm?


>
> Thanks.
> S:
>
>
>
>
>
>
>
> >
> > > Thanks,
> > > S.
> > > ___
> > > ffmpeg-user mailing list
> > > ffmpeg-user@ffmpeg.org
> > > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > >
> > > To unsubscribe, visit link above, or email
> > > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
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> >
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Re: [FFmpeg-user] Same output bit depth and sampling frequency as input?

[Paul B Mahol]

On Fri, Feb 12, 2021 at 12:48 AM Marco Mircoli  wrote:

> Hello everybody,
>  wondering how to set the same output audio specs (sample rate anche bit
> depth) as input.
>
> I tried like that
>
> ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
> 5:TP=-1.5:LRA=7 -sample_fmt s16 -ar 44100 FILE_TO
>
>
>
> if I have a 48KHZ input, the output is 44100, so doesn't work for my needs
>
> Tried without
>
>
> ffmpeg -i FILE_FROM -af dynaudnorm,loudnorm=I=-16.
> 5:TP=-1.5:LRA=7  FILE_TO
>
> The result is: same bit depth of input file but 192Khz sample rate. (file
> input sr is 48khz)
>
> Anybody can give me a suggest?
>

Use shell scripting.

Also using those 2 filters at same time is entirely invalid and pointless.



> Thanks,
> S.
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Re: [FFmpeg-user] Is there something about inputting raw frames

[Paul B Mahol]

On Fri, Feb 12, 2021 at 9:36 AM Mark Filipak (ffmpeg) 
wrote:

> On 02/12/2021 02:48 AM, pdr0 wrote:
> > Mark Filipak (ffmpeg) wrote
> >> On 02/12/2021 02:28 AM, pdr0 wrote:
> >>> Mark Filipak (ffmpeg) wrote
>  On 02/12/2021 01:27 AM, pdr0 wrote:
> > Mark Filipak (ffmpeg) wrote
> >> Is there something about inputting raw frames that I don't know?
> >>
> >> I'm using 'vspipe' to pipe raw frames to 'ffmpeg -i pipe:'.
> >> The vapoursynth script, 'Mark's.vpy', is known good.
> >> The output of vapoursynth is known good.
> >> I've tried to be careful to retain valid PTSs, but apparently have
> >> failed.
> >> The output should be around 1200 frames, but 364 frames are dropped.
> >> I've frame stepped through the target, 'Mark's_script_6.mkv', and
> the
> >> frames that are there are in
> >> the correct order.
> >> The only thing I can guess is that ffmpeg handles 48/1.001fps raw
> >> video
> >> frames in such a way that
> >> PTS is not valid or can't be changed with 'setpts=fps=6/1001'.
> >> Can anyone see an error. Or, lacking an error, does anyone know of a
> >> workaround?
> >>
> >> Thanks.
> >>
> >> Mark's_script_6.cmd
> >> =
> >> ECHO from vapoursynth import core>Mark's.vpy
> >> ECHO video =
> >> core.ffms2.Source(source='Mark\'s_source.mkv')>>Mark's.vpy
> >> ECHO import havsfunc as havsfunc>>Mark's.vpy
> >> ECHO video = havsfunc.InterFrame(video, Preset="medium",
> >> Tuning="smooth",
> >> InputType="2D",
> >> NewNum=48000, NewDen=1001, GPU=True)>>Mark's.vpy
> >> ECHO video.set_output()>>Mark's.vpy
> >> vspipe --y4m Mark's.vpy - | ffmpeg -thread_queue_size 2048 -i pipe:
> >> -filter_complex
> >> "setpts=N*1001/6/TB, split[1][2], [1]shuffleframes=0 1 2 3 3,
> >> select=not(eq(mod(n\,5)\,4))[3],
> >> [2]tblend=all_expr='if(eq(mod(X,2),mod(Y,2)),TOP,BOTTOM)',
> >> shuffleframes=0
> >> 1 2 2 3,
> >> select=eq(mod(n\,5)\,4)[4], [3][4]interleave" -i Mark's_source.mkv
> >> -map
> >> 0:v -map 1:a -codec:v
> >> libx265 -x265-params "crf=16:qcomp=0.60" -codec:a copy -codec:s copy
> >> Mark's_script_6.mkv -y
> >
> >
> > Are you trying to keep the same frames from vapoursynth output node,
> > but
> > assign 6/1001 fps and timestamps instead of 48000/1001 ?
> > (effectively
> > making it a speed up)
> >
> > If so, the workaround is : add after the Interframe line
> >
> > video = core.std.AssumeFPS(video, fpsnum=6, fpsden=1001)
> 
>  After your suggested addition to the python script, Mark's.vpy,
>  With '-filter_complex "setpts=N*1001/6/TB, split[1][2]...' there
> are
>  335 drops.
>  With '-filter_complex "split[1][2]...' there are 190 drops.
> >>>
> >>> The workaround is correct for the PTS
> >>>
> >>>AssumeFPS is used to change the frame rate (and timestamps) without
> >>> changing the frame count. It just assigns a framerate (and their PTS).
> So
> >>> you would use that instead of setpts
> >>>
> >>> There are no frame drops or additions from vapoursynth. The framecount,
> >>> framerate and PTS are correct at that point. You can verify this by
> >>> encoding
> >>> the vpy script directly without other filters.
> >>>
> >>> So this implies the some drops are from setpts, and some other drops
> are
> >>> from some of your other filters
> >>
> >> Well, I previously removed the 'setpts' directives with no change.
> Also, I
> >> previously tested with no
> >> ffmpeg filters at all and got the expected sped up video. I honestly
> can't
> >> see anything else to
> >> discover. But I'll start stripping filters one by one and hack a
> solution
> >> (or make a discovery). Of
> >> course, the 'shuffleframes' directives are most suspect, but I've used
> >> 'shuffleframes' in the past
> >> and succeeded.
> >>
> >> Thanks for your help. It was instrumental. The rest is up to me.
> >
> >
> >
> >
> > I realize this is ffmpeg-user board, but why not do some of the video
> > processing in vapoursynth ? You're already using it for part of it.  But
> I'm
> > not sure what you're trying to do ?
> >
> > It looks like you're taking the 48000/1001 interpolated frame doubled
> output
> > and adding a "BC~C.bc~c" frame to every group of 4, to make up the
> > 6/1001?
> >
> > What is a "BC~C.bc~c" frame ? Is it that checkerboard blend?
> > http://ffmpeg.org/pipermail/ffmpeg-user/2021-February/051852.html
> >
> > Either way, cadence wise that's going to be worse in terms of smoothness
> > then an optical flow retimed 6/1001 . (Some people would argue it's
> > worse period, you're retiming it and making it look like a soap opera...)
>
> You know, I think that "soap opera" opinion is entirely bogus. In a movie
> theater, projectors are
> triple shuttered. That essentially brings the frame rate up to 72fps with
> picture rate of 24pps (or,
> if you include the times 

Re: [FFmpeg-user] exposure filter

[Paul B Mahol]

On Thu, Feb 11, 2021 at 8:15 AM Michael Koch 
wrote:

> Am 11.02.2021 um 00:50 schrieb Paul B Mahol:
> > On Thu, Feb 11, 2021 at 12:43 AM Paul B Mahol  wrote:
> >
> >>
> >> On Wed, Feb 10, 2021 at 11:51 PM Michael Koch <
> astroelectro...@t-online.de>
> >> wrote:
> >>
> >>> Am 10.02.2021 um 23:17 schrieb Paul B Mahol:
> >>>> Why you do not start fixing your pdf file, it have many obsolete and
> >>> even
> >>>> false statements.
> >>> What's wrong and how can it be improved?
> >>>
> >>>
> >> Almost everything is wrong about it.
> >>
> >> One single example, acrossover filter have examples and you claim it
> does
> >> not.
> >>
> > Also, you looks to have sole authority to call shufflepixels filter
> > useless, and couple others.
>
> Paul is referring to chapter 2.149 "FFmpeg: Suggestions for improvement"
> in my book
> http://www.astro-electronic.de/FFmpeg_Book.pdf
>
> It's possible that some of the suggestions have already been realized in
> the meantime, but I may have overlooked them. In such cases please just
> let me know, and I'll be happy to change that in my document.
>
>
I do not have time for consuming review of your text dump.
Hire some professional writers instead.

I do not have enough time to develop code, review patches, answer user
questions and also to fight some text that is not updated at all.
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Re: [FFmpeg-user] astats to csv file

[Paul B Mahol]

On Thu, Feb 11, 2021 at 4:40 PM Benjamin Houtman 
wrote:

> Thank you,
> I ran the
>
> ffprobe -f lavfi -i amovie=X.flac,astats=metadata=1 -show_frames -of csv
>
> command, but I don't need the entire printout, just the "Parsed_astats..."
> data that appears at the end of the printout. How would I limit the results
> to just that?
>
>
It is not written as example but with english skills you can find that
command is:
ffprobe -f lavfi -i amovie=INPUT.flac,astats=metadata=1 -show_entries
tags=lavfi.astats.Overall.Noise_floor_count


> On Thu, Feb 11, 2021 at 10:07 AM Michael Koch  >
> wrote:
>
> > Am 11.02.2021 um 15:48 schrieb Benjamin Houtman:
> > > How would that translate into a csv file?
> >
> > The example in chapter 2.113 of my book might help. It's for another
> > filter, but also with *.csv output.
> > http://www.astro-electronic.de/FFmpeg_Book.pdf
> >
> > Michael
> >
> > ___
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> >
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>
>
>
> --
> Benjamin Houtman
> Archivist
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Re: [FFmpeg-user] astats to csv file

[Paul B Mahol]

On Thu, Feb 11, 2021 at 2:35 PM Benjamin Houtman 
wrote:

> I've tried ffprobe at a basic level but it doesn't provide what I'm looking
> for, at least as I know how to use it. I basically want the
> "Parsed_astats..."  outputs in columns as header: value, for multiple wav
> files in a folder.
>

ffprobe -f lavfi -i amovie=X.flac,astats=metadata=1 -show_frames


> On Thu, Feb 11, 2021 at 8:16 AM Paul B Mahol  wrote:
>
> > On Thu, Feb 11, 2021 at 2:13 PM Benjamin Houtman 
> > wrote:
> >
> > > I'm trying to find a simple way to push a file's astats metadata to a
> csv
> > > file with headers. It feels like it should be easy to do, but I've not
> > been
> > > successful so far.
> > >
> >
> > Tried ffprobe csv output with filtering?
> >
> >
> > > Thanks
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Re: [FFmpeg-user] astats to csv file

[Paul B Mahol]

On Thu, Feb 11, 2021 at 2:13 PM Benjamin Houtman 
wrote:

> I'm trying to find a simple way to push a file's astats metadata to a csv
> file with headers. It feels like it should be easy to do, but I've not been
> successful so far.
>

Tried ffprobe csv output with filtering?


> Thanks
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Re: [FFmpeg-user] exposure filter

[Paul B Mahol]

On Thu, Feb 11, 2021 at 12:43 AM Paul B Mahol  wrote:

>
>
> On Wed, Feb 10, 2021 at 11:51 PM Michael Koch 
> wrote:
>
>> Am 10.02.2021 um 23:17 schrieb Paul B Mahol:
>> > Why you do not start fixing your pdf file, it have many obsolete and
>> even
>> > false statements.
>>
>> What's wrong and how can it be improved?
>>
>>
> Almost everything is wrong about it.
>
> One single example, acrossover filter have examples and you claim it does
> not.
>

Also, you looks to have sole authority to call shufflepixels filter
useless, and couple others.

>
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Re: [FFmpeg-user] exposure filter

[Paul B Mahol]

On Wed, Feb 10, 2021 at 11:51 PM Michael Koch 
wrote:

> Am 10.02.2021 um 23:17 schrieb Paul B Mahol:
> > Why you do not start fixing your pdf file, it have many obsolete and even
> > false statements.
>
> What's wrong and how can it be improved?
>
>
Almost everything is wrong about it.

One single example, acrossover filter have examples and you claim it does
not.

Michael
>
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Re: [FFmpeg-user] exposure filter

[Paul B Mahol]

On Thu, Feb 11, 2021 at 12:10 AM Michael Koch 
wrote:

> Am 10.02.2021 um 23:28 schrieb Paul B Mahol:
> > On Wed, Feb 10, 2021 at 11:17 PM Paul B Mahol  wrote:
> >
> >> Why you do not start fixing your pdf file, it have many obsolete and
> even
> >> false statements.
> >>
> > Also to add to that, you already have limiter filter if you need to limit
> > something, no point in doing it in this filter.
>
> Yes it's right that the limiter is easy to use after a filter. But for
> the user it's important to know which filters have a built-in limiter
> and which filters don't have it. This should be documented for each filter.
>

Nope. You clearly have big knowledge gap and have no authority to write
what should be documented.


> Michael
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Re: [FFmpeg-user] exposure filter

[Paul B Mahol]

On Wed, Feb 10, 2021 at 11:17 PM Paul B Mahol  wrote:

>
> Why you do not start fixing your pdf file, it have many obsolete and even
> false statements.
>

Also to add to that, you already have limiter filter if you need to limit
something, no point in doing it in this filter.

It is easy and dangerous to criticize something with completely lack of
knowledge.
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Re: [FFmpeg-user] exposure filter

[Paul B Mahol]

Why you do not start fixing your pdf file, it have many obsolete and even
false statements.
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Re: [FFmpeg-user] exposure filter

[Paul B Mahol]

On Wed, Feb 10, 2021 at 8:14 PM Michael Koch 
wrote:

> Hello,
>
> in the new "exposure" filter it's unclear in which order the operations
> are applied. First exposure, then black level? Or first black level,
> then exposure? Does the filter have a limiter if the result is out of
> range? These infos should be added to the documentation.
>


At same time. Doubt that internal stuff is interesting to average joe,
filter works with floats only. Limiter is not needed.




>
> Michael
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Re: [FFmpeg-user] h264 encoding of bgr images into .mp4 file with libav

[Paul B Mahol]

On Wed, Feb 10, 2021 at 12:55 PM laddoe  wrote:

> I'm trying to encode(h264) a series of .png into a mp4 file. A cv::Mat
> holds the png data (BGR) and that is converted to YUV420P which is then
> encoded and written to a .mp4 file. I have added a block statement in the
> code to store image on the disk (before encoding) and that image is correct
> so the frame holds the correct data before it gets passed. The
> avcodec_send_frame returns 0 so up to that point everything works. But I
> get an mp4 file of 1 MB and I can't open it with vlc. Below is a short
> summary of my code
>
>
> ecodec.h
>
> class ECodec
> {
> public:
>
> MovieCodec();
> ~MovieCodec();
> void MatToFrame( cv::Mat& image );
> void encode( AVFrame *frame, AVPacket *pkt );
>
> private:
>
> FILE* m_file;
> AVCodec* m_encoder = NULL;
> AVCodecContext* m_codecContextOut = NULL;
> AVPacket* m_packet = NULL;
>
> };
>
>
> ecodec.cpp
>
> ECodec::ECodec() :
> //m_encoder( avcodec_find_encoder_by_name( videoCodec.c_str()))
> m_encoder( avcodec_find_encoder( AV_CODEC_ID_H264 ))
> {
> m_file = fopen( "c:\\tmp\\outputVideo.mp4", "wb");
> }
>
>
> void ECodec::MatToFrame( cv::Mat& image )
> {
> int ret( 0 );
> int frameRate( 24 );
> AVFrame *frame = NULL;
>
> m_encoder( avcodec_find_encoder( AV_CODEC_ID_H264 ))
> m_codecContextOut = avcodec_alloc_context3( m_encoder );
>
> m_codecContextOut->width = 800;
> m_codecContextOut->height = 640;
> m_codecContextOut->bit_rate = 40;//m_codecContextOut->width *
> m_codecContextOut->height * 3;
> m_codecContextOut->time_base = (AVRational){1, 24};
> m_codecContextOut->framerate = (AVRational){24, 1};
> m_codecContextOut->codec_tag = AV_CODEC_ID_H264;
> m_codecContextOut->pix_fmt = AV_PIX_FMT_YUV420P;
> m_codecContextOut->codec_type = AVMEDIA_TYPE_VIDEO;
> m_codecContextOut->gop_size = 1;
> m_codecContextOut->max_b_frames = 1;
>
> av_log_set_level(AV_LOG_VERBOSE);
>
> ret = av_opt_set(m_codecContextOut->priv_data, "preset", "slow", 0);
>
> ret = avcodec_open2(m_codecContextOut, m_encoder, NULL);
>
> frame = av_frame_alloc();
>
> frame->format = AV_PIX_FMT_YUV420P;
> frame->width = image.cols();
> frame->height = image.rows();
>
>
> ret = av_image_alloc(frame->data, frame->linesize, frame->width,
> frame->height, AV_PIX_FMT_YUV420P, 1);
>
> if (ret < 0)
> {
> return;
> }
>
> struct SwsContext *sws_ctx;
> sws_ctx = sws_getContext((int)image.cols(), (int)image.rows(),
> AV_PIX_FMT_RGB24,
>  (int)image.cols(), (int)image.rows(),
> AV_PIX_FMT_YUV420P,
>  0, NULL, NULL, NULL);
>
> const uint8_t* rgbData[1] = { (uint8_t* )image.getData() };
> int rgbLineSize[1] = { 3 * image.cols() };
>
> sws_scale(sws_ctx, rgbData, rgbLineSize, 0, image.rows(), frame->data,
> frame->linesize);
>
> frame->pict_type = AV_PICTURE_TYPE_I;
>
> cv::Mat yuv420p(frame->height + frame->height/2, frame->width,
> CV_8UC1,frame->data[0]);
> cv::Mat cvmIm;
> cv::cvtColor(yuv420p,cvmIm,CV_YUV420p2BGR);
> cv::imwrite("c:\\tmp\\rawimage.png", cvmIm);
> //OK
>
> m_packet = av_packet_alloc();
> ret = av_new_packet( m_packet, m_codecContextOut->width *
> m_codecContextOut->height * 3 );
>
> /* encode the image */
> encode( frame, m_packet );
>
>
> avcodec_free_context(_codecContextOut);
> av_frame_free();
> av_packet_free( _packet );
> }
>
>
> void ECodec::encode( AVFrame *frame, AVPacket *pkt )
> {
> int ret;
> /* send the frame to the encoder */
> ret = avcodec_send_frame( m_codecContextOut, frame);
>
> if (ret < 0)
> {
> fprintf(stderr, "Error sending a frame for encoding\n");
> exit(1);
> }
>
> do
> {
> ret = avcodec_receive_packet(m_codecContextOut, pkt);
> if (ret == 0)
> {
> fwrite(pkt->data, 1, pkt->size, m_file );
>

You are writing raw encoded frames here.

You nowhere use mov muxer code for muxing.

av_packet_unref(pkt);
>
> break;
> }
> else if ((ret < 0) && (ret != AVERROR(EAGAIN)))
> {
> return;
> }
> else if (ret == AVERROR(EAGAIN))
> {
>  ret = avcodec_send_frame(m_codecContextOut, NULL);
>  if (0 > ret)
>  {
>  return;
>  }
> }
> } while (ret == 0);
> }
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Re: [FFmpeg-user] List etiquette [was: Re: alternating checkerboard]

[Paul B Mahol]

On Wed, Feb 10, 2021 at 1:26 AM Reino Wijnsma  wrote:

> On 2021-02-09T18:19:27+0100, Carl Eugen Hoyos  wrote:
> > Am Di., 9. Feb. 2021 um 18:06 Uhr schrieb Carl Zwanzig :
> >
> > [...]
> >
> > Please stop this.
> Double standards, Carl?
>
> You're always quick to warn people to not top-post on this mailing list
> (and rightly so!), but now that it concerns a developer here it's not
> important that also he respects the rules/etiquette? I'm baffled!
>
> I fully agree with Carl Zwanzig. The rules apply to all users.
>

I'm not user.


> --
> Reino
>
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Re: [FFmpeg-user] alternating checkerboard

[Paul B Mahol]

On Tue, Feb 9, 2021 at 10:02 AM Paul B Mahol  wrote:

>
>
> On Tue, Feb 9, 2021 at 9:45 AM Mark Filipak (ffmpeg) 
> wrote:
>
>> On 02/09/2021 03:14 AM, Michael Koch wrote:
>> > Am 09.02.2021 um 04:49 schrieb Mark Filipak (ffmpeg):
>> >> I don't know the details of the filters that work on pixels ('blend',
>> for example). I hope a
>> >> knowledgeable person will respond with suggestions.
>> >>
>> >> I'd like to experiment with a filter that mixes pixels from 2
>> progressive streams on an
>> >> alternating checkerboard pattern. Got any suggestions?
>> >>  ___
>> >> |# # # # # # # # #   | % % % % % % % %|#%#%#%#%#%#%#%#%#
>> >> | # # # # # # # #|% % % % % % % % %   |%#%#%#%#%#%#%#%#%
>> >> |# # # # # # # # # + | % % % % % % % %  = |#%#%#%#%#%#%#%#%#
>> >> | # # # # # # # #|% % % % % % % % %   |%#%#%#%#%#%#%#%#%
>> >> |# # # # # # # # #   | % % % % % % % %|#%#%#%#%#%#%#%#%#
>> >>
>> >>
>> >
>> > Have a look at the "maskedmerge" filter. In your case the mergemap file
>> would contain a pattern of
>> > 0's and 255's. This file can be generated with "geq" filter (see the
>> example in chapter 2.107).
>>
>> Thanks for the suggestion. Actually, it appears that the 'blend' filter
>> does what I want, but I'm
>> still testing to be sure.
>> blend=all_expr='if(eq(mod(X,2),mod(Y,2)),TOP,BOTTOM)'
>> 'blend' appears to input 2 frames in 2 streams and output 1 frame (with
>> unknown PTS). 'tblend', on
>> the other hand, appears to take 2 frames in 1 stream and output 2 frames,
>> not 1 frame, (again, with
>> unknown PTS). It sure would be nice if these filter functions were
>> adequately documented, especially
>> regarding how they determine output PTS. But you shouldn't listen to me
>> because, to quote the
>> authorities: "That makes no sense".
>>
>>
> Why you write unknown PTS? What you are talking about, please do not be
> dense.
>
> PTS is always deterministic from these filters.
>

ffmpeg -i ~/Videos/red-leaf-tips.avi -i ~/Videos/musical-box.avi -lavfi
format=yuv444p,framepack=format=frameseq,stereo3d=in=al:out=chl,setsar=1/2
-f nut -c:v rawvideo -|mpv -

It is not exactly what you want, as this variant do not discard data. But
you get an idea.

>
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Re: [FFmpeg-user] List etiquette [was: Re: alternating checkerboard]

[Paul B Mahol]

Top posting is allowed and is healthy habit.
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Re: [FFmpeg-user] alternating checkerboard

[Paul B Mahol]

On Tue, Feb 9, 2021 at 9:45 AM Mark Filipak (ffmpeg) 
wrote:

> On 02/09/2021 03:14 AM, Michael Koch wrote:
> > Am 09.02.2021 um 04:49 schrieb Mark Filipak (ffmpeg):
> >> I don't know the details of the filters that work on pixels ('blend',
> for example). I hope a
> >> knowledgeable person will respond with suggestions.
> >>
> >> I'd like to experiment with a filter that mixes pixels from 2
> progressive streams on an
> >> alternating checkerboard pattern. Got any suggestions?
> >>  ___
> >> |# # # # # # # # #   | % % % % % % % %|#%#%#%#%#%#%#%#%#
> >> | # # # # # # # #|% % % % % % % % %   |%#%#%#%#%#%#%#%#%
> >> |# # # # # # # # # + | % % % % % % % %  = |#%#%#%#%#%#%#%#%#
> >> | # # # # # # # #|% % % % % % % % %   |%#%#%#%#%#%#%#%#%
> >> |# # # # # # # # #   | % % % % % % % %|#%#%#%#%#%#%#%#%#
> >>
> >>
> >
> > Have a look at the "maskedmerge" filter. In your case the mergemap file
> would contain a pattern of
> > 0's and 255's. This file can be generated with "geq" filter (see the
> example in chapter 2.107).
>
> Thanks for the suggestion. Actually, it appears that the 'blend' filter
> does what I want, but I'm
> still testing to be sure.
> blend=all_expr='if(eq(mod(X,2),mod(Y,2)),TOP,BOTTOM)'
> 'blend' appears to input 2 frames in 2 streams and output 1 frame (with
> unknown PTS). 'tblend', on
> the other hand, appears to take 2 frames in 1 stream and output 2 frames,
> not 1 frame, (again, with
> unknown PTS). It sure would be nice if these filter functions were
> adequately documented, especially
> regarding how they determine output PTS. But you shouldn't listen to me
> because, to quote the
> authorities: "That makes no sense".
>
>
Why you write unknown PTS? What you are talking about, please do not be
dense.

PTS is always deterministic from these filters.



> > Michael
> >
> > ___
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> >
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>
>
> --
> I don't have a dog.
> And furthermore, my dog doesn't bite.
> And furthermore, you provoked him.
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Re: [FFmpeg-user] alternating checkerboard

[Paul B Mahol]

Also I wanted to help you, but you constant bullying of project discourages
me.

On Tue, Feb 9, 2021 at 10:02 AM Paul B Mahol  wrote:

>
>
> On Tue, Feb 9, 2021 at 9:45 AM Mark Filipak (ffmpeg) 
> wrote:
>
>> On 02/09/2021 03:14 AM, Michael Koch wrote:
>> > Am 09.02.2021 um 04:49 schrieb Mark Filipak (ffmpeg):
>> >> I don't know the details of the filters that work on pixels ('blend',
>> for example). I hope a
>> >> knowledgeable person will respond with suggestions.
>> >>
>> >> I'd like to experiment with a filter that mixes pixels from 2
>> progressive streams on an
>> >> alternating checkerboard pattern. Got any suggestions?
>> >>  ___
>> >> |# # # # # # # # #   | % % % % % % % %|#%#%#%#%#%#%#%#%#
>> >> | # # # # # # # #|% % % % % % % % %   |%#%#%#%#%#%#%#%#%
>> >> |# # # # # # # # # + | % % % % % % % %  = |#%#%#%#%#%#%#%#%#
>> >> | # # # # # # # #|% % % % % % % % %   |%#%#%#%#%#%#%#%#%
>> >> |# # # # # # # # #   | % % % % % % % %|#%#%#%#%#%#%#%#%#
>> >>
>> >>
>> >
>> > Have a look at the "maskedmerge" filter. In your case the mergemap file
>> would contain a pattern of
>> > 0's and 255's. This file can be generated with "geq" filter (see the
>> example in chapter 2.107).
>>
>> Thanks for the suggestion. Actually, it appears that the 'blend' filter
>> does what I want, but I'm
>> still testing to be sure.
>> blend=all_expr='if(eq(mod(X,2),mod(Y,2)),TOP,BOTTOM)'
>> 'blend' appears to input 2 frames in 2 streams and output 1 frame (with
>> unknown PTS). 'tblend', on
>> the other hand, appears to take 2 frames in 1 stream and output 2 frames,
>> not 1 frame, (again, with
>> unknown PTS). It sure would be nice if these filter functions were
>> adequately documented, especially
>> regarding how they determine output PTS. But you shouldn't listen to me
>> because, to quote the
>> authorities: "That makes no sense".
>>
>>
> Why you write unknown PTS? What you are talking about, please do not be
> dense.
>
> PTS is always deterministic from these filters.
>
>
>
>> > Michael
>> >
>> > ___
>> > ffmpeg-user mailing list
>> > ffmpeg-user@ffmpeg.org
>> > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
>> >
>> > To unsubscribe, visit link above, or email
>> > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>>
>>
>> --
>> I don't have a dog.
>> And furthermore, my dog doesn't bite.
>> And furthermore, you provoked him.
>> ___
>> ffmpeg-user mailing list
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>> https://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>> To unsubscribe, visit link above, or email
>> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>
>
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

This is because it does not allow arbitrary values, just some small set.

On Sun, Feb 7, 2021 at 3:51 PM Paul B Mahol  wrote:

>
>
> On Sun, Feb 7, 2021 at 3:49 PM Thibeau 
> wrote:
>
>> It is very strange that if you change the value you set comment from
>> 'Cover
>> (front)' to e.g 'Cover' it will set the tag to cover but ffmpeg won't
>> recognize it and display it as Other any ideas why this is?
>>
>
> Because if your set value does not match any from specification section I
> linked previously
> it will pick first entry, And first entry is "Other".
>
>
>>
>> On Sun, 7 Feb 2021 at 11:03, Thibeau 
>> wrote:
>>
>> > First ffprobe shows before executing ffmpeg command . I installed the
>> git
>> > version for ffmpeg to make sure it was the same as yours. It seems
>> you're
>> > infact corrrect and it allows for you to set the tag to another and
>> value
>> > and will default to Other. Thx for the help of explaining it. Didn't
>> find
>> > the docs on my own so thx for giving a link to them.
>> > Don't know alot about ffmpeg either but adding -vsync 2 to your command
>> > would seem like a good idea no? Than the error in the command (see
>> second
>> > command [2]) would go away. Unless I'm mistaken and it's intent to be
>> not
>> > use vsync 2 method.
>> > -- [1]
>> > ffprobe '29 My Girl.flac'
>> > ffprobe version N-100998-g134193a2ff Copyright (c) 2007-2021 the FFmpeg
>> > developers
>> >   built with gcc 9.3.0 (Gentoo 9.3.0-r2 p4)
>> >   configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64
>> > --docdir=/usr/share/doc/ffmpeg-/html --mandir=/usr/share/man
>> > --enable-shared --cc=x86_64-pc-linux-gnu-gcc
>> --cxx=x86_64-pc-linux-gnu-g++
>> > --ar=x86_64-pc-linux-gnu-ar --nm=x86_64-pc-linux-gnu-nm
>> > --ranlib=x86_64-pc-linux-gnu-ranlib
>> > --pkg-config=x86_64-pc-linux-gnu-pkg-config --optflags='-march=native
>> -O3
>> > -pipe -flto=5 -fgraphite-identity -floop-nest-optimize -ftree-vectorize
>> > -fsched-pressure -fipa-pta -fno-semantic-interposition
>> -fuse-linker-plugin
>> > -fno-common -fdevirtualize-at-ltrans -falign-functions=32
>> > --param=l1-cache-size=32 --param=l1-cache-line-size=64
>> > --param=l2-cache-size=6144' --disable-static --enable-avfilter
>> > --enable-avresample --disable-stripping --disable-optimizations
>> > --disable-libcelt --enable-nonfree --disable-indev=v4l2
>> > --disable-outdev=v4l2 --disable-indev=oss --disable-indev=jack
>> > --disable-outdev=oss --enable-bzlib --disable-runtime-cpudetect
>> > --disable-debug --disable-gcrypt --enable-gnutls --disable-gmp
>> --enable-gpl
>> > --disable-hardcoded-tables --enable-iconv --disable-libtls
>> > --disable-libxml2 --disable-lzma --disable-network --disable-opencl
>> > --enable-openssl --enable-postproc --disable-libsmbclient
>> --disable-ffplay
>> > --disable-sdl2 --disable-vaapi --disable-vdpau --disable-vulkan
>> > --enable-xlib --enable-libxcb --enable-libxcb-shm --enable-libxcb-xfixes
>> > --enable-zlib --disable-libcdio --disable-libiec61883
>> --disable-libdc1394
>> > --enable-libcaca --disable-openal --disable-opengl --disable-libv4l2
>> > --disable-libpulse --enable-libdrm --disable-libjack
>> > --disable-libopencore-amrwb --disable-libopencore-amrnb
>> --disable-libcodec2
>> > --enable-libdav1d --disable-libfdk-aac --disable-libopenjpeg
>> > --enable-libbluray --disable-libgme --disable-libgsm
>> --disable-libaribb24
>> > --disable-mmal --disable-libmodplug --enable-libopus --disable-libilbc
>> > --disable-librtmp --disable-libssh --disable-libspeex --disable-libsrt
>> > --enable-librsvg --disable-ffnvcodec --enable-libvorbis --disable-libvpx
>> > --disable-libzvbi --disable-appkit --disable-libbs2b
>> --disable-chromaprint
>> > --disable-cuda-llvm --disable-libflite
>> > --disable-frei0r --disable-libfribidi --enable-fontconfig
>> --disable-ladspa
>> > --disable-libass --disable-libtesseract --disable-lv2
>> --enable-libfreetype
>> > --disable-libvidstab --disable-librubberband --disable-libzmq
>> > --disable-libzimg --disable-libsoxr --enable-pthreads
>> > --disable-libvo-amrwbenc --enable-libmp3lame --disable-libkvazaar
>> > --disable-libaom --disable-libopenh264 --disable-librav1e
>> > --disable-libsnappy --disable-libtheora --disable-libtwolame
>> > --disable-libwebp --disable-li

Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

sable-htmlpages --enable-manpages
> >   libavutil  56. 64.100 / 56. 64.100
> >   libavcodec 58.121.100 / 58.121.100
> >   libavformat58. 67.100 / 58. 67.100
> >   libavdevice58. 11.103 / 58. 11.103
> >   libavfilter 7.102.100 /  7.102.100
> >   libavresample   4.  0.  0 /  4.  0.  0
> >   libswscale  5.  8.100 /  5.  8.100
> >   libswresample   3.  8.100 /  3.  8.100
> >   libpostproc55.  8.100 / 55.  8.100
> > Input #0, flac, from '29 My Girl.flac':
> >   Metadata:
> > ALBUM   : Prodigal Son: Season 1 (Original Television
> > Soundtrack)
> > ARTIST  : Nathaniel Blume
> > DATE: 2020
> > TITLE   : My Girl
> > track   : 29
> >   Duration: 00:02:55.60, start: 0.00, bitrate: 1424 kb/s
> >   Stream #0:0: Audio: flac, 44100 Hz, stereo, s32 (24 bit)
> >   Stream #0:1: Video: mjpeg (Baseline), yuvj444p(pc,
> > bt470bg/unknown/unknown), 640x640 [SAR 118:118 DAR 1:1], 90k
> > tbr, 90k tbn, 90k tbc (attached pic)
> > Metadata:
> >   comment : Other
> > Stream mapping:
> >   Stream #0:1 -> #0:0 (mjpeg (native) -> png (native))
> >   Stream #0:0 -> #0:1 (flac (native) -> flac (native))
> > Press [q] to stop, [?] for help
> > [swscaler @ 0xa8fd5d00] deprecated pixel format used, make sure you
> > did set range correctly
> > [flac @ 0xa8f532c0] Frame rate very high for a muxer not efficiently
> > supporting it.
> > Please consider specifying a lower framerate, a different muxer or
> -vsync 2
> > Output #0, flac, to 'out.flac':
> >   Metadata:
> > ALBUM   : Prodigal Son: Season 1 (Original Television
> > Soundtrack)
> > ARTIST  : Nathaniel Blume
> > DATE: 2020
> > TITLE   : My Girl
> > track   : 29
> > encoder : Lavf58.67.100
> >   Stream #0:0: Video: png, rgb24(pc, bt470bg/unknown/unknown,
> > progressive), 640x640 [SAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k fps, 90k
> tbn
> > (attached pic)
> > Metadata:
> >   comment : Cover (front)
> >   encoder : Lavc58.121.100 png
> >   Stream #0:1: Audio: flac, 44100 Hz, stereo, s32 (24 bit), 128 kb/s
> > Metadata:
> >   encoder : Lavc58.121.100 flac
> > frame=1 fps=0.0 q=-0.0 Lsize=   31013kB time=00:02:55.60
> > bitrate=1446.8kbits/s speed= 359x
> > video:565kB audio:30440kB subtitle:0kB other streams:0kB global
> > headers:0kB muxing overhead: 0.026684%
> > - [1]
> >
> > On Sun, 7 Feb 2021 at 10:35, Paul B Mahol  wrote:
> >
> >> On Sun, Feb 7, 2021 at 11:19 AM Paul B Mahol  wrote:
> >>
> >> >
> >> >
> >> > On Sun, Feb 7, 2021 at 12:59 AM MediaMouth 
> wrote:
> >> >
> >> >> On Feb 6, 2021, at 3:32 PM, Paul B Mahol  wrote:
> >> >> >
> >> >> > Can you share file?
> >> >> Yes.  The O.P. made it available here: https://ufile.io/bgunj34e <
> >> >> https://ufile.io/bgunj34e>
> >> >>
> >> >> >
> >> >> > ffmpeg -i INPUT.png -i AUDIO.flac -map:v 0 -metadata comment=Other
> >> >> > -disposition:v:0 attached_pic -map:a 1 -metadata album=test
> >> FINAL.flac
> >> >> >
> >> >> > Note that metadata order is important. The command above adds
> comment
> >> >> > metadata to video stream.
> >> >>
> >> >> Thanks for posting.  Clear and helpful.  Unfortunately, I got an
> error
> >> >> Here are the steps I did...
> >> >>
> >> >
> >> > Your steps are all invalid. Why you put first audio then video?
> >> > My command only works with first video stream and after that audio
> >> stream.
> >> > In ffmpeg every item position matters.
> >> >
> >>
> >> Also you can not remove description of attached picture at all. "Other"
> is
> >> just default value if no other valid values are set.
> >> You can change value to "Cover (front)" if you like, please read 4.14
> >> section of https://id3.org/id3v2.4.0-frames
> >>
> >> So command to change Other to Cover is:
> >> ffmpeg -i INPUT.flac -c:v png -disposition:v:1 attached_pic
> >> -metadata:s:v:0
> >> comment="Cover (front)" OUTPUT.flac
> >>
> >>
> >>
> >> >
> >> >
> >> >> ffmpeg -i '29 My Girl.flac' -an -vcodec copy '29 My Girl.png';
> #Extract
> >> >> the album art to an external file.
> >> >> ffmpeg -i '29 My Girl.flac' -map 0 -map -0:v:0 -c copy '29 My
> >> >> Girl-NoArt.flac'; #Remove the album art, save results to a new .flac
> >> file.
> >> >> ffmpeg -i '29 My Girl-NoArt.flac' -i '29 My Girl.png' -map:v 0
> >> -metadata
> >> >> comment=SomethingOtherThanOther -disposition:v:0 attached_pic -map:a
> 1
> >> >> -metadata comment= '29 My Girl-ReArt.flac'; #Add the album art back
> >> using
> >> >> the metadata commands as you suggest, and store to yet another .flac
> >> file.
> >> >> Error received:  [swscaler @ 0x7f9648038000] deprecated pixel format
> >> >> used, make sure you did set range correctly
> >> >>
> >> >>
> >> >> ___
> >> >> ffmpeg-user mailing list
> >> >> ffmpeg-user@ffmpeg.org
> >> >> https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >> >>
> >> >> To unsubscribe, visit link above, or email
> >> >> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
> >> >
> >> >
> >> ___
> >> ffmpeg-user mailing list
> >> ffmpeg-user@ffmpeg.org
> >> https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >>
> >> To unsubscribe, visit link above, or email
> >> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
> >
> >
> ___
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>
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

On Sun, Feb 7, 2021 at 11:31 AM Thibeau 
wrote:

> I did that command you gave with video and than audio in that order
> checked it three times.
> The metadata went to the audio not the video.
> Feel free to test it yourself with the provided file.
> Please let me know if it was my mistake and the command is correct when
> you execute it.
>
>
I can not guess what you typed. And user assume to much things these days.

So next time post full ffmpeg log.



> > On 7 Feb 2021, at 11:19, Paul B Mahol  wrote:
> >
> > On Sun, Feb 7, 2021 at 12:59 AM MediaMouth  wrote:
> >
> >>> On Feb 6, 2021, at 3:32 PM, Paul B Mahol  wrote:
> >>>
> >>> Can you share file?
> >> Yes.  The O.P. made it available here: https://ufile.io/bgunj34e <
> >> https://ufile.io/bgunj34e>
> >>
> >>>
> >>> ffmpeg -i INPUT.png -i AUDIO.flac -map:v 0 -metadata comment=Other
> >>> -disposition:v:0 attached_pic -map:a 1 -metadata album=test FINAL.flac
> >>>
> >>> Note that metadata order is important. The command above adds comment
> >>> metadata to video stream.
> >>
> >> Thanks for posting.  Clear and helpful.  Unfortunately, I got an error
> >> Here are the steps I did...
> >>
> >
> > Your steps are all invalid. Why you put first audio then video?
> > My command only works with first video stream and after that audio
> stream.
> > In ffmpeg every item position matters.
> >
> >
> >> ffmpeg -i '29 My Girl.flac' -an -vcodec copy '29 My Girl.png'; #Extract
> >> the album art to an external file.
> >> ffmpeg -i '29 My Girl.flac' -map 0 -map -0:v:0 -c copy '29 My
> >> Girl-NoArt.flac'; #Remove the album art, save results to a new .flac
> file.
> >> ffmpeg -i '29 My Girl-NoArt.flac' -i '29 My Girl.png' -map:v 0 -metadata
> >> comment=SomethingOtherThanOther -disposition:v:0 attached_pic -map:a 1
> >> -metadata comment= '29 My Girl-ReArt.flac'; #Add the album art back
> using
> >> the metadata commands as you suggest, and store to yet another .flac
> file.
> >> Error received:  [swscaler @ 0x7f9648038000] deprecated pixel format
> used,
> >> make sure you did set range correctly
> >>
> >>
> >> ___
> >> ffmpeg-user mailing list
> >> ffmpeg-user@ffmpeg.org
> >> https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >>
> >> To unsubscribe, visit link above, or email
> >> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
> > ___
> > ffmpeg-user mailing list
> > ffmpeg-user@ffmpeg.org
> > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> > To unsubscribe, visit link above, or email
> > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
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>
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

On Sun, Feb 7, 2021 at 11:19 AM Paul B Mahol  wrote:

>
>
> On Sun, Feb 7, 2021 at 12:59 AM MediaMouth  wrote:
>
>> On Feb 6, 2021, at 3:32 PM, Paul B Mahol  wrote:
>> >
>> > Can you share file?
>> Yes.  The O.P. made it available here: https://ufile.io/bgunj34e <
>> https://ufile.io/bgunj34e>
>>
>> >
>> > ffmpeg -i INPUT.png -i AUDIO.flac -map:v 0 -metadata comment=Other
>> > -disposition:v:0 attached_pic -map:a 1 -metadata album=test FINAL.flac
>> >
>> > Note that metadata order is important. The command above adds comment
>> > metadata to video stream.
>>
>> Thanks for posting.  Clear and helpful.  Unfortunately, I got an error
>> Here are the steps I did...
>>
>
> Your steps are all invalid. Why you put first audio then video?
> My command only works with first video stream and after that audio stream.
> In ffmpeg every item position matters.
>

Also you can not remove description of attached picture at all. "Other" is
just default value if no other valid values are set.
You can change value to "Cover (front)" if you like, please read 4.14
section of https://id3.org/id3v2.4.0-frames

So command to change Other to Cover is:
ffmpeg -i INPUT.flac -c:v png -disposition:v:1 attached_pic -metadata:s:v:0
comment="Cover (front)" OUTPUT.flac



>
>
>> ffmpeg -i '29 My Girl.flac' -an -vcodec copy '29 My Girl.png'; #Extract
>> the album art to an external file.
>> ffmpeg -i '29 My Girl.flac' -map 0 -map -0:v:0 -c copy '29 My
>> Girl-NoArt.flac'; #Remove the album art, save results to a new .flac file.
>> ffmpeg -i '29 My Girl-NoArt.flac' -i '29 My Girl.png' -map:v 0 -metadata
>> comment=SomethingOtherThanOther -disposition:v:0 attached_pic -map:a 1
>> -metadata comment= '29 My Girl-ReArt.flac'; #Add the album art back using
>> the metadata commands as you suggest, and store to yet another .flac file.
>> Error received:  [swscaler @ 0x7f9648038000] deprecated pixel format
>> used, make sure you did set range correctly
>>
>>
>> ___
>> ffmpeg-user mailing list
>> ffmpeg-user@ffmpeg.org
>> https://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>> To unsubscribe, visit link above, or email
>> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>
>
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

On Sun, Feb 7, 2021 at 12:59 AM MediaMouth  wrote:

> On Feb 6, 2021, at 3:32 PM, Paul B Mahol  wrote:
> >
> > Can you share file?
> Yes.  The O.P. made it available here: https://ufile.io/bgunj34e <
> https://ufile.io/bgunj34e>
>
> >
> > ffmpeg -i INPUT.png -i AUDIO.flac -map:v 0 -metadata comment=Other
> > -disposition:v:0 attached_pic -map:a 1 -metadata album=test FINAL.flac
> >
> > Note that metadata order is important. The command above adds comment
> > metadata to video stream.
>
> Thanks for posting.  Clear and helpful.  Unfortunately, I got an error
> Here are the steps I did...
>

Your steps are all invalid. Why you put first audio then video?
My command only works with first video stream and after that audio stream.
In ffmpeg every item position matters.


> ffmpeg -i '29 My Girl.flac' -an -vcodec copy '29 My Girl.png'; #Extract
> the album art to an external file.
> ffmpeg -i '29 My Girl.flac' -map 0 -map -0:v:0 -c copy '29 My
> Girl-NoArt.flac'; #Remove the album art, save results to a new .flac file.
> ffmpeg -i '29 My Girl-NoArt.flac' -i '29 My Girl.png' -map:v 0 -metadata
> comment=SomethingOtherThanOther -disposition:v:0 attached_pic -map:a 1
> -metadata comment= '29 My Girl-ReArt.flac'; #Add the album art back using
> the metadata commands as you suggest, and store to yet another .flac file.
> Error received:  [swscaler @ 0x7f9648038000] deprecated pixel format used,
> make sure you did set range correctly
>
>
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

On Sun, Feb 7, 2021 at 12:23 AM MediaMouth  wrote:

> On Feb 6, 2021, at 15:13, Paul B Mahol  wrote:
> >
> > I take random flac with cover art and converted it to flac with ffmpeg,
> > cover art is kept.
>
> Is there any way at all to get control over the per stream metadata --
> specifically the 1st video stream which seems to be the one holding the art?
>
> And would you know why it contains, by default, the key/value pair
> "comment:Other"?
>
>
Can you share file?

ffmpeg -i INPUT.png -i AUDIO.flac -map:v 0 -metadata comment=Other
-disposition:v:0 attached_pic -map:a 1 -metadata album=test FINAL.flac

Note that metadata order is important. The command above adds comment
metadata to video stream.



> Thanks.
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

Here is full command:

ffmpeg -i COVER.png -i AUDIO.flac -map:v 0 -disposition:v:0 attached_pic
-map:a 1 FINAL.flac

On Sat, Feb 6, 2021 at 6:01 PM Thibeau 
wrote:

> >Ugh.  Always coming back as "comment=other"?
> Yea, but I'm not sure the value of tag is other and not just a value ffmpeg
> doesn't  recognise and theirfor defaults to showing Other.
>
> > I'm hoping one of the FFmpeg devs can shed some light on this
> Same, I'm a real noob at ffmpeg. Trying to learn it but it's really hard
> with the limited time I have to do it.
>
> >That would be great.  Thanks.  I haven't found one that works for .flac
> yet.
> Disclaimer the commands are all for the file I uploaded if it's another
> file don't forget to rename it in the commands.
>
> > ffmpeg -i '29 My Girl.flac' -an -vcodec copy '29 My Girl.png'
>
> This removes the art along with the associated metadata (comment=other)
> > ffmpeg -i '29 My Girl.flac' -map 0 -map -0:v:0 -c copy '29 My
> Girl-NoArt.flac'
>
> Reattach the image:
> ffmpeg -i 29\ My\ Girl-NoArt.flac -i 29\ My\ Girl.png -map 1 -map 0 -c copy
> -disposition:0 attached_pic out.flac (refrence:
>
> https://stackoverflow.com/questions/54717175/how-do-i-add-a-custom-thumbnail-to-a-mp4-file-using-ffmpeg
> )
>
> Hope this helps!
>
>
>
> On Sat, 6 Feb 2021 at 16:34, MediaMouth  wrote:
>
> >
> > lOn Feb 6, 2021, at 07:44, Thibeau 
> wrote:
> > >
> > > I indeed figured out how to extract the cover art than remove it from
> > the song.
> > >
> > > I already know how to place it back too. The problem is I don’t know
> > what the metadata comment tag is.
> > > I can’t find it on the image after extracting with exiftools or any
> > other tool for that matter.
> > I saw the same. It seems to be metadata associated with that video
> stream.
> >
> >
> > > If I reattach the cover afterwards with ffmpeg, the metadata tag is
> > still there.
> > Ugh.  Always coming back as "comment=other"?
> >
> > > I guess it must be stored somewhere, but I have no idea how to acces or
> > change
> > I'm hoping one of the FFmpeg devs can shed some light on this
> >
> >
> > >
> > > P.S the commands I used to split/extract/... just came from
> > stackoverflow answers, happy to send to commands here if you need them.
> > That would be great.  Thanks.  I haven't found one that works for .flac
> > yet.
> >
> >
> > ___
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> > ffmpeg-user@ffmpeg.org
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> >
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> > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

On Sun, Feb 7, 2021 at 12:12 AM Paul B Mahol  wrote:

>
>
> On Sat, Feb 6, 2021 at 3:52 PM Harvey Pikelberger 
> wrote:
>
>>
>>
>> > On Feb 6, 2021, at 02:01, Paul B Mahol  wrote:
>> >
>> > Despite other incorrect information. FFmpeg can handle album art for
>> .flac
>> > file just fine.
>>
>> Paul, thanks for the correction.
>> I was able to extract the art to a separate file and also remove the art,
>> which in turn removed the associated metadata. (Thibeau, I'll send you
>> those commands momentarily.  Paul may improve on it.  He knows this stuff
>> far better)
>>
>> Paul, are you saying FFmpeg can add album art to .flac files?  I seems
>> possible do so with mp3 manipulating id3 info, but how to do so with
>> .flac?Thanks
>>
>>
> I take random flac with cover art and converted it to flac with ffmpeg,
> cover art is kept.
>


Also, note that I use git version, not any release version but git version.


>
>
>>
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

On Sat, Feb 6, 2021 at 3:52 PM Harvey Pikelberger 
wrote:

>
>
> > On Feb 6, 2021, at 02:01, Paul B Mahol  wrote:
> >
> > Despite other incorrect information. FFmpeg can handle album art for
> .flac
> > file just fine.
>
> Paul, thanks for the correction.
> I was able to extract the art to a separate file and also remove the art,
> which in turn removed the associated metadata. (Thibeau, I'll send you
> those commands momentarily.  Paul may improve on it.  He knows this stuff
> far better)
>
> Paul, are you saying FFmpeg can add album art to .flac files?  I seems
> possible do so with mp3 manipulating id3 info, but how to do so with
> .flac?Thanks
>
>
I take random flac with cover art and converted it to flac with ffmpeg,
cover art is kept.


>
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Re: [FFmpeg-user] Smooth Streaming demuxer ?

[Paul B Mahol]

How much are you informed about subject you ask?

On Sat, Feb 6, 2021 at 12:40 PM mtv123--- via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

> Hi
> It seems FFmpeg have a muxer only for Smooth Streaming, does anyone know
> if a demuxer is worked on or will be added in future ?
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Re: [FFmpeg-user] Removing a metadata tag from a specfic stream

[Paul B Mahol]

Despite other incorrect information. FFmpeg can handle album art for .flac
file just fine.

It just can not alter metadata of attached art file IIRC.

On Sat, Feb 6, 2021 at 10:29 AM Thibeau 
wrote:

> Yea I suppose I could just remove the art entirely and that works, but
> could ffmpeg get the art out of the songs? I wouldn’t know how to get them
> and out. But if it’s possible I might be able to just reattach them
> afterwards and see if the tag hasn’t gone or maybe edit while the image is
> separated.
>
> > On 6 Feb 2021, at 00:56, MediaMouth  wrote:
> >
> > On Feb 5, 2021, at 12:21 PM, MediaMouth  wrote:
> >> The obvious approach (targeting it as the first video track) didn't
> work:
> >>> ffmpeg -y -i '29 My Girl.flac' -metadata:s:v:0 comment= '29 My
> GirlB.flac'
> >>> ffprobe '29 My GirlB.flac'
> >
> > One possibility would be to simply remove the album art (you could save
> it externally beforehand).
> > That will, in turn, get rid of the associated metadata "comment". I
> tested it.  It works.
> > From what I understand ffmpeg (and for that matter Adobe) can't handle
> album art for .flac files.
> >
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Re: [FFmpeg-user] Can ffmpeg interface/use SVPflow?

[Paul B Mahol]

On Fri, Feb 5, 2021 at 11:31 PM Mark Filipak (ffmpeg) 
wrote:

> On 02/05/2021 04:55 PM, Paul B Mahol wrote:
> > On Fri, Feb 5, 2021 at 10:54 PM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 02/05/2021 04:43 PM, Paul B Mahol wrote:
> >>> On Fri, Feb 5, 2021 at 10:21 PM Mark Filipak (ffmpeg) <
> >> markfili...@bog.us>
> >>> wrote:
> >>>
> >>>> On 02/05/2021 04:13 PM, Paul B Mahol wrote:
> >>>>> On Fri, Feb 5, 2021 at 10:05 PM Mark Filipak (ffmpeg) <
> >>>> markfili...@bog.us>
> >>>>> wrote:
> >>>>>
> >>>>>> On 02/05/2021 03:43 PM, Carl Eugen Hoyos wrote:
> >>>>>>> Am Fr., 5. Feb. 2021 um 21:00 Uhr schrieb Mark Filipak (ffmpeg)
> >>>>>>> :
> >>>>>>>>
> >>>>>>>> On 02/05/2021 02:22 PM, Carl Eugen Hoyos wrote:
> >>>>>>>>> Am Fr., 5. Feb. 2021 um 20:19 Uhr schrieb Mark Filipak (ffmpeg)
> >>>>>>>>> :
> >>>>>>>>>>
> >>>>>>>>>> Can ffmpeg interface/use SVPflow?
> >>>>>>>>>
> >>>>>>>>> Please read a little about OpenSource software and
> >>>>>>>>> please understand that OpenSource has nothing to
> >>>>>>>>> do with "no payment" or whatever you may be thinking
> >>>>>>>>> of.
> >>>>>>>
> >>>>>>>> I'm not sure why you have responded that way.
> >>>>>>>>
> >>>>>>>> To the best of my knowledge, for what it's worth, SVPflow is
> >>>>>>>> open source. SVPflow2 is closed source.
> >>>>>>>
> >>>>>>> You didn't provide a link and I only found a proprietary
> >>>>>>> library.
> >>>>>>> If there is an open-source variant that is better than
> >>>>>>> minterpolate: Patch probably welcome!
> >>>>>>
> >>>>>> Does that mean the answer to my question: "Can ffmpeg interface/use
> >>>>>> SVPflow?", is No?
> >>>>>>
> >>>>>
> >>>>> That information is strictly confidential.
> >>>>
> >>>> Hahaha... Do you see my signature line below, Paul? That applies to
> you
> >>>> guys.
> >>>>
> >>>>
> >>> If you know how to setup VS or AVS you would know.
> >>
> >> What is "VS" and "AVS", Paul? What would I know? What the hell are you
> >> 'talking' about?
> >>
> >
> > Ask your new friends on doom9 forum.
> >
> > VS - Vapoursynth
> > AVS - AVIsynth
>
> My new friends? Paul, are you eating magic mushrooms?
>
> No, I don't know how to set up VS or AVS.
>

Their documentation is best one ever. You will learn everything in couple
of hours.


>
> >>>> --
> >>>> I don't have a dog.
> >>>> And furthermore, my dog doesn't bite.
> >>>> And furthermore, you provoked him.
> >>> --
> >>> I don't have a dog.
> >>> And furthermore, my dog doesn't bite.
> >>> And furthermore, you provoked him.--
> >> -- >> I don't have a dog.
> >> And furthermore, my dog doesn't bite.
> >> And furthermore, you provoked him.
> > --
> > I don't have a dog.
> > And furthermore, my dog doesn't bite.
> > And furthermore, you provoked him.
> --
> I don't have a dog.
> And furthermore, my dog doesn't bite.
> And furthermore, you provoked him.
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Re: [FFmpeg-user] Can ffmpeg interface/use SVPflow?

[Paul B Mahol]

On Fri, Feb 5, 2021 at 10:54 PM Mark Filipak (ffmpeg) 
wrote:

> On 02/05/2021 04:43 PM, Paul B Mahol wrote:
> > On Fri, Feb 5, 2021 at 10:21 PM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 02/05/2021 04:13 PM, Paul B Mahol wrote:
> >>> On Fri, Feb 5, 2021 at 10:05 PM Mark Filipak (ffmpeg) <
> >> markfili...@bog.us>
> >>> wrote:
> >>>
> >>>> On 02/05/2021 03:43 PM, Carl Eugen Hoyos wrote:
> >>>>> Am Fr., 5. Feb. 2021 um 21:00 Uhr schrieb Mark Filipak (ffmpeg)
> >>>>> :
> >>>>>>
> >>>>>> On 02/05/2021 02:22 PM, Carl Eugen Hoyos wrote:
> >>>>>>> Am Fr., 5. Feb. 2021 um 20:19 Uhr schrieb Mark Filipak (ffmpeg)
> >>>>>>> :
> >>>>>>>>
> >>>>>>>> Can ffmpeg interface/use SVPflow?
> >>>>>>>
> >>>>>>> Please read a little about OpenSource software and
> >>>>>>> please understand that OpenSource has nothing to
> >>>>>>> do with "no payment" or whatever you may be thinking
> >>>>>>> of.
> >>>>>
> >>>>>> I'm not sure why you have responded that way.
> >>>>>>
> >>>>>> To the best of my knowledge, for what it's worth, SVPflow is
> >>>>>> open source. SVPflow2 is closed source.
> >>>>>
> >>>>> You didn't provide a link and I only found a proprietary
> >>>>> library.
> >>>>> If there is an open-source variant that is better than
> >>>>> minterpolate: Patch probably welcome!
> >>>>
> >>>> Does that mean the answer to my question: "Can ffmpeg interface/use
> >>>> SVPflow?", is No?
> >>>>
> >>>
> >>> That information is strictly confidential.
> >>
> >> Hahaha... Do you see my signature line below, Paul? That applies to you
> >> guys.
> >>
> >>
> > If you know how to setup VS or AVS you would know.
>
> What is "VS" and "AVS", Paul? What would I know? What the hell are you
> 'talking' about?
>

Ask your new friends on doom9 forum.

VS - Vapoursynth
AVS - AVIsynth


>
> >> --
> >> I don't have a dog.
> >> And furthermore, my dog doesn't bite.
> >> And furthermore, you provoked him.
> > --
> > I don't have a dog.
> > And furthermore, my dog doesn't bite.
> > And furthermore, you provoked him.--
> I don't have a dog.
> And furthermore, my dog doesn't bite.
> And furthermore, you provoked him.
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Re: [FFmpeg-user] Can ffmpeg interface/use SVPflow?

[Paul B Mahol]

On Fri, Feb 5, 2021 at 10:21 PM Mark Filipak (ffmpeg) 
wrote:

> On 02/05/2021 04:13 PM, Paul B Mahol wrote:
> > On Fri, Feb 5, 2021 at 10:05 PM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 02/05/2021 03:43 PM, Carl Eugen Hoyos wrote:
> >>> Am Fr., 5. Feb. 2021 um 21:00 Uhr schrieb Mark Filipak (ffmpeg)
> >>> :
> >>>>
> >>>> On 02/05/2021 02:22 PM, Carl Eugen Hoyos wrote:
> >>>>> Am Fr., 5. Feb. 2021 um 20:19 Uhr schrieb Mark Filipak (ffmpeg)
> >>>>> :
> >>>>>>
> >>>>>> Can ffmpeg interface/use SVPflow?
> >>>>>
> >>>>> Please read a little about OpenSource software and
> >>>>> please understand that OpenSource has nothing to
> >>>>> do with "no payment" or whatever you may be thinking
> >>>>> of.
> >>>
> >>>> I'm not sure why you have responded that way.
> >>>>
> >>>> To the best of my knowledge, for what it's worth, SVPflow is
> >>>> open source. SVPflow2 is closed source.
> >>>
> >>> You didn't provide a link and I only found a proprietary
> >>> library.
> >>> If there is an open-source variant that is better than
> >>> minterpolate: Patch probably welcome!
> >>
> >> Does that mean the answer to my question: "Can ffmpeg interface/use
> >> SVPflow?", is No?
> >>
> >
> > That information is strictly confidential.
>
> Hahaha... Do you see my signature line below, Paul? That applies to you
> guys.
>
>
If you know how to setup VS or AVS you would know.


> --
> I don't have a dog.
> And furthermore, my dog doesn't bite.
> And furthermore, you provoked him.
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Re: [FFmpeg-user] Can ffmpeg interface/use SVPflow?

[Paul B Mahol]

On Fri, Feb 5, 2021 at 10:05 PM Mark Filipak (ffmpeg) 
wrote:

> On 02/05/2021 03:43 PM, Carl Eugen Hoyos wrote:
> > Am Fr., 5. Feb. 2021 um 21:00 Uhr schrieb Mark Filipak (ffmpeg)
> > :
> >>
> >> On 02/05/2021 02:22 PM, Carl Eugen Hoyos wrote:
> >>> Am Fr., 5. Feb. 2021 um 20:19 Uhr schrieb Mark Filipak (ffmpeg)
> >>> :
> 
>  Can ffmpeg interface/use SVPflow?
> >>>
> >>> Please read a little about OpenSource software and
> >>> please understand that OpenSource has nothing to
> >>> do with "no payment" or whatever you may be thinking
> >>> of.
> >
> >> I'm not sure why you have responded that way.
> >>
> >> To the best of my knowledge, for what it's worth, SVPflow is
> >> open source. SVPflow2 is closed source.
> >
> > You didn't provide a link and I only found a proprietary
> > library.
> > If there is an open-source variant that is better than
> > minterpolate: Patch probably welcome!
>
> Does that mean the answer to my question: "Can ffmpeg interface/use
> SVPflow?", is No?
>

That information is strictly confidential.


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> And furthermore, my dog doesn't bite.
> And furthermore, you provoked him.
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Re: [FFmpeg-user] Out of virtual memory (swap, I assume) = telecine+setpts combination

[Paul B Mahol]

On Thu, Feb 4, 2021 at 11:02 PM Mark Filipak (ffmpeg) 
wrote:

> On 02/04/2021 04:50 PM, Paul B Mahol wrote:
> > Well, perhaps your command queues too much frame and that is bad.
>
> Perhaps?
>
>
Perhaps not. I just checked telecine+setpts with your arguments and it
leaks 0 bytes.


> ffmpeg -i source.mkv -map 0 -filter_complex "telecine,
> setpts=N*1001/3/TB" -codec:v libx265
> -x265-params crf=20:qcomp=0.60 -codec:a copy -codec:s copy "with both
> telecine and setpts.mkv"
>
> What does "...perhaps your command queues too much frame" mean?
>
>
> > On Thu, Feb 4, 2021 at 10:34 PM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 02/03/2021 12:23 AM, Carl Zwanzig wrote:
> >>> On 2/2/2021 2:52 PM, Mark Filipak (ffmpeg) wrote:
> >>>> If that's true, what's eating the swap? Transcoding parts 1, 2, 3, & 4
> >> all succeeded when running
> >>>> concurrently. Transcoding part 5 fails, even when ffmpeg is the only
> >> app running.
> >>>
> >>> Out of curiosity, have you tried looking at the file with an mkv
> >> analyzer like MediaInfo, Elecard's
> >>> analyzer* or maybe another one? (Is there metadata towards the end of
> >> the file containing junk info?)
> >>>
> >>> How does it behave if you drop the minterpolate or one of the other
> >> filters, output to a null file,
> >>> or use a different output codec? (Anything to try isolating the
> problem.)
> >>
> >> If rising memory allocation and rising swap commit indicates a memory
> >> leak, then the telecine filter
> >> has a memory leak.
> >>
> >> **WRONG**
> >>
> >> OHMYGOD! The problem isn't telecine. The problem isn't setpts. The
> problem
> >> is telecine + setpts in
> >> combination!
> >>
> >> **CORRECTION**
> >> If rising memory allocation and rising swap commit indicates a memory
> >> leak, then the combination of
> >> telecine and setpts filters provokes a memory leak.
> >>
> >> My trouble packet is 'telecine+setpts memory leak.zip'. I can't put it
> in
> >> dropbox because dropbox
> >> has now added a 100 MB limit in an attempt to force me to pay.
> >>
> >> 'telecine+setpts memory leak.zip' is 328 MiB.
> >>
> >> With no way to get 'telecine+setpts memory leak.zip' into the
> developers'
> >> hands, I guess all my
> >> efforts will go to naught.
> >> ___
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> >>
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> >
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> >
>
>
> --
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> And furthermore, my dog doesn't bite.
> And furthermore, you provoked him.
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Re: [FFmpeg-user] Out of virtual memory (swap, I assume) = telecine+setpts combination

[Paul B Mahol]

Well, perhaps your command queues too much frame and that is bad.

On Thu, Feb 4, 2021 at 10:34 PM Mark Filipak (ffmpeg) 
wrote:

> On 02/03/2021 12:23 AM, Carl Zwanzig wrote:
> > On 2/2/2021 2:52 PM, Mark Filipak (ffmpeg) wrote:
> >> If that's true, what's eating the swap? Transcoding parts 1, 2, 3, & 4
> all succeeded when running
> >> concurrently. Transcoding part 5 fails, even when ffmpeg is the only
> app running.
> >
> > Out of curiosity, have you tried looking at the file with an mkv
> analyzer like MediaInfo, Elecard's
> > analyzer* or maybe another one? (Is there metadata towards the end of
> the file containing junk info?)
> >
> > How does it behave if you drop the minterpolate or one of the other
> filters, output to a null file,
> > or use a different output codec? (Anything to try isolating the problem.)
>
> If rising memory allocation and rising swap commit indicates a memory
> leak, then the telecine filter
> has a memory leak.
>
> **WRONG**
>
> OHMYGOD! The problem isn't telecine. The problem isn't setpts. The problem
> is telecine + setpts in
> combination!
>
> **CORRECTION**
> If rising memory allocation and rising swap commit indicates a memory
> leak, then the combination of
> telecine and setpts filters provokes a memory leak.
>
> My trouble packet is 'telecine+setpts memory leak.zip'. I can't put it in
> dropbox because dropbox
> has now added a 100 MB limit in an attempt to force me to pay.
>
> 'telecine+setpts memory leak.zip' is 328 MiB.
>
> With no way to get 'telecine+setpts memory leak.zip' into the developers'
> hands, I guess all my
> efforts will go to naught.
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Re: [FFmpeg-user] Filter just the 1st audio channel of the input.

[Paul B Mahol]

On Wed, Feb 3, 2021 at 7:48 PM Nicolas George  wrote:

> Marco Mircoli (12021-02-03):
> >wandering if it is possible to process just the 1st audio channel of
> an
> > input.
> > I don't know how many channels I will have as input, but I know that I
> have
> > to process just the 1st and the others I have not to considerate.
> >
> > INPUT from mono to multichannel
> > filters.
> > OUTPUT just one channel (mono)
> >
> > I've tried this but doesn't works
> >
> > ffmpeg -i FILE_FROM -map_channel 0.0.0 -af
> > highpass=f=70,adeclick,loudnorm=I=-16.5:TP=-1.5:LRA=7 FILE_TO
>
> Just use pan=mono|c0=c0 to extract the channel you want wherever you
> want in the graph.
>
> No need to completely split the channels, it was meant for versions that
> lack the pan filter.
>


You missed fact that user needs all channels again, not just first.


>
> Regards,
>
> --
>   Nicolas George
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Re: [FFmpeg-user] Filter just the 1st audio channel of the input.

[Paul B Mahol]

On Wed, Feb 3, 2021 at 2:05 PM Marco Mircoli  wrote:

> Hello,
>wandering if it is possible to process just the 1st audio channel of an
> input.
> I don't know how many channels I will have as input, but I know that I have
> to process just the 1st and the others I have not to considerate.
>
> INPUT from mono to multichannel
> filters.
> OUTPUT just one channel (mono)
>
> I've tried this but doesn't works
>
> ffmpeg -i FILE_FROM -map_channel 0.0.0 -af
> highpass=f=70,adeclick,loudnorm=I=-16.5:TP=-1.5:LRA=7 FILE_TO
>

you will need to extract each channel by channelsplit filter, and later
merge it with other filters.
This makes filtergraph description longer.
Some filters support processing only specific channel but most do not as it
does not make sense for such filters.


>
> Any suggest?
> Thanks.
> S.
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Re: [FFmpeg-user] Out of virtual memory (swap, I assume)

[Paul B Mahol]

On Tue, Feb 2, 2021 at 11:20 PM Carl Eugen Hoyos  wrote:

> Am Di., 2. Feb. 2021 um 19:38 Uhr schrieb Paul B Mahol :
> >
> > minterpolate filter may leak memory, i havent tried to run it under
> > valgrind because i have only 4GB of RAM, and slow CPU.
>
> valgrind shows no leak, afair minterpolate only allocates its
> buffers once at initialization.
>

It may still leak frames.


> Carl Eugen
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Re: [FFmpeg-user] Out of virtual memory (swap, I assume)

[Paul B Mahol]

minterpolate filter may leak memory, i havent tried to run it under
valgrind because i have only 4GB of RAM, and slow CPU.

On Tue, Feb 2, 2021 at 7:22 PM Mark Filipak (ffmpeg) 
wrote:

> Windows 10-1803
> Memory: 32GB
> Virtual memory: 128GB on an SSD-RAID0.
>
> No other applications running.
>
> ffmpeg -i "THE LAST EMPEROR [1987] source part5.mkv" -map 0
> -filter_complex
> "minterpolate=fps=48000/1001:mi_mode=mci:mc_mode=obmc:scd=fdiff:scd_threshold=10:vsbmc=1:search_param=32,
>
> telecine=pattern=3322,pp=linblenddeint, setpts=N*1001/6/TB" -codec:v
> libx265 -x265-params
> "crf=20:qcomp=0.60" -codec:a copy -codec:s copy "THE LAST EMPEROR [1987]
> part5.mkv"
>
> As the transcode proceeds, I watch the virtual memory commit climb
> steadily.
> As the transcode proceeds, I watch the memory allocation top out and then
> hover there.
> I went to bed, but I assume that when the virtual memory commit tops out,
> ffmpeg reports the
> following error:
> "Error while filtering: Cannot allocate memory
> "Failed to inject frame into filter network: Cannot allocate memory
> "Error while processing the decoded data for stream #0:0
> "Conversion failed!"
>
> There's something about 'source part5.mkv' that causes the fault, but I'll
> be damned if I can find it.
>
> I tried the part 5 transcode 3 times:
> 1, While concurrently transcoding parts 1-4 (which all succeeded), and
> 2, On its own, and finally
> 3, I again extracted part 5 via MKVToolNIX, and transcoded again.
> That last time I made sure that 'source part5.mkv' played correctly (with
> correct running times in
> MPV) and with monotonically increasing timestamps before transcoding for
> the 3rd time.
>
> I took 4 pertinent files, renamed them, zipped them as part5.zip
>  ...6,138,036,350 bytes.
> Inside part5.zip is:
> [part5.zip]\source part5.mkv   ...6,150,347,006 bytes
> [part5.zip]\source part5.mkv timestamps.txt   ...5,095,698 bytes
> [part5.zip]\part5.mkv failure #3.log   ...3,372,428 bytes
> [part5.zip]\part5.mkv timestamps.txt   ...3,114,801 bytes
>
> *** Well, dropbox seems to be malfunctioning. Can you suggest an
> alternative? ***
>
> Any/all ideas regarding how I can successfully transcode part 5 will be
> much appreciated even if
> it/they fail.
>
> Thanks and Good Health. Regards,
> Mark.
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Re: [FFmpeg-user] minterpolate problem

[Paul B Mahol]

On Mon, Feb 1, 2021 at 3:49 AM Mark Filipak (ffmpeg) 
wrote:

> Developers, kindly pay attention to this post...
>
> On 01/31/2021 10:24 AM, Phil Rhodes via ffmpeg-user wrote:
> >   On Sunday, 31 January 2021, 13:24:39 GMT, Rodney Baker <
> rodney.ba...@iinet.net.au> wrote:
> >
> >> See man ffmpeg for detailed description of the options... > try typing
> "man ffmpeg" into a google search and see what comes up... >
> http://manpages.org/ffmpeg...> Look for the section titled "Generic
> Options". > There you'll find the detailed descriptions you're looking for.
>
> ISSUE #1:
> I did not get this reply from Rodney Baker. The list server apparently is
> not sending out all
> messages to all user-ffmpeg subscribers. I've noticed this before but
> didn't 'say' anything.
>
> > Oh, right, yes, obviously. That's what I call first-order retrievability!
> > To the broader question:
> > Experts posting on this list frequently become irritated when people ask
> basic questions. Well, actually, they tend to become almost comically
> abusive, but that's another issue.
> > If these people want to field less basic questions, they should improve
> the documentation.
> > If they don't want to improve the documentation, they should let someone
> else do it. People have repeatedly offered.
> > Rejecting all of those options and returning to the abuse is not
> reasonable.
> > P
>
> ISSUE #2:
> ffmpeg -h says: "See man ffmpeg for detailed description of the options."
> So I went into my Linux VM, submitted "man ffmpeg", and redirected it into
> a text file.
> 'man ffmpeg' is as uninformative as 'ffmpeg -h' -- this is typical of
> 'man' pages.
>
> I had no idea 'http://manpages.org/ffmpeg' existed until I read Phil's
> response to Rodney's reply.
>
> Why is ffmpeg documentation so scattered?
> Why is ffmpeg documentation so disorganized?
> Why is ffmpeg documentation so vague?
> Why is the English grammar in ffmpeg documentation so poor?
>
> Regarding the wiki:
> It is mostly just an empty shell.
> Based on my experience with Wikipedia, /wiki.*/gi=='amateurish',
> /wiki.*/gi=='ad hoc',
> /wiki.*/gi=='incomplete', /wiki.*/gi=='non authoritative',
> /wiki.*/gi=='untrustworthy',
> /wiki.*/gi=='disinformation'.
> Question: Why would any authoritative person contribute to a wiki? Answer:
> They don't bother.
>
> I've been running ffmpeg approximately 14 hours per day for over 1/2 year.
> I don't have a life
> outside ffmpeg. (I haven't watched popular TV since 1983 and have no
> social life.) I'm 74 years old
> and may not have much more time left (except I do take good care of
> myself).
>
>
People your age enjoy their life. You should too. Why you want to dealt
with digital video...



> I'm a university engineer. I know video. I know a little about ffmpeg. The
> stumbling block is ffmpeg
> documentation.
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Re: [FFmpeg-user] minterpolate ...for Paul

[Paul B Mahol]

On Sat, Jan 30, 2021 at 11:46 AM Phil Rhodes via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

>
>
> On Saturday, 30 January 2021, 08:24:10 GMT, Moritz Barsnick <
> barsn...@gmx.net> wrote:
>  > Please don't throw dirt around here. Carl Eugen is trying to help.
>
> I think it's worth being clear that it sometimes isn't very easy to tell
> whether Carl Eugen is trying to help.
>

Censor yourself out.


> P
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Re: [FFmpeg-user] Next Release

[Paul B Mahol]

On Fri, Jan 29, 2021 at 9:19 PM Reindl Harald 
wrote:

>
>
> Am 29.01.21 um 20:21 schrieb Armin Hasitzka:
> >>> For release versions, one just make sure that FATE report is not
> yellow.
> >>
> >> Alright, that sounds simple enough;  CI testing for the win \o/
> >>
> >> Much appreciated + thanks for your hard work <3
> >
> > So I've been running FATE on current `master` all the way back to
> > `3c700c82cbee49d1f40b6a7063f4b084ae3ee747` and I can't seem to get far
> > with it, one of these two always fail:
> >
> > - ./tests/ref/fate/filter-earwax
> > - ./tests/ref/lavf/mxf
> >
> > Are these tests flaky, am I doing something wrong or is current
> > `master` broken? Happy to share more setup details if it helps
> > (running on Ubuntu 16.04.7)
>
> well, but current HEAD is always perfect *lol+
>

It is. And you are unable to grasp that.

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Re: [FFmpeg-user] minterpolate ...for Paul

[Paul B Mahol]

Hello administrators, please ban this corporate evangelist from here,
thanks.

On Fri, Jan 29, 2021 at 4:02 PM Phil Rhodes via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

>  >You can technically, but minterpolate is not very user friendly - It's
> too> slow for real work and feedback, and you cannot keyframe the settings
> on> different scenes very easily. It's barely usable unless you program
> your own> GUI around libavfilter
> Yes - this is somewhere that commandline tools tend to start to become
> less suitable, when you can't really audition settings and evaluate
> tradeoffs very easily.
> If you have a problem you need to solve quickly, there is optical flow
> interpolation in the free version of Resolve, with better optical flow
> interpolation in the paid version, which might make life a bit easier. Much
> of Resolve is multi-threaded and GPU-accelerated (I'm not completely sure
> if the optical flow is, to be honest, but I'd be astounded if it wasn't)
> and you can to tweak things visually for the best possible compromise.
> Naturally if you're just tweaking around with ffmpeg for fun, fine.
> P
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Re: [FFmpeg-user] VOICE POST PRODUCTION IN ONE LINE

[Paul B Mahol]

tializing filters!
> Failed to inject frame into filter network: Invalid argument
> Error while processing the decoded data for stream #0:0
> [AVIOContext @ 0x555cfa976140] Statistics: 0 seeks, 0 writeouts
> [AVIOContext @ 0x555cfa97c880] Statistics: 98432 bytes read, 2 seeks
> Conversion failed!
>
> Il giorno ven 29 gen 2021 alle ore 13:02 Paul B Mahol 
> ha
> scritto:
>
> > On Fri, Jan 29, 2021 at 12:52 PM Chris Miceli 
> wrote:
> >
> > > If you wish to have output from your service, you would need to check
> the
> > > documentation of the library you are using to execute the shell
> command.
> > > They often have the ability to capture stdout and stderr which is what
> > you
> > > need.
> > >
> > > Often, these libraries may interpret shell commands in different ways
> and
> > > so it may be parsing the ffmpeg command line (this is something to keep
> > in
> > > mind, not necessarily what is happening).
> > >
> > > If you can run your command from the shell normally, there is likely
> > > something in your service code which is causing it.
> > >
> >
> > The only thing that come to mind, is that version is old that it does not
> > have that specific filter.
> > Also note that those audio filters are not magic way to solve your
> problems
> > by putting them in filtergraph and forgetting about them next second.
> >
> >
> > > On Fri, Jan 29, 2021, 10:21 PM Paul B Mahol  wrote:
> > >
> > > > On Fri, Jan 29, 2021 at 12:18 PM Marco Mircoli  >
> > > > wrote:
> > > >
> > > > > Thanks,
> > > > > is there a way to understand if ffmpeg has aborted the process?
> > > > > I'm using a Linux version for an online service.
> > > > > It would be nice to let ffmpeg write a log file when there is a
> > problem
> > > > > (process aborted)
> > > > > How to do it?
> > > > >
> > > >
> > > > By providing full ffmpeg output. (Not output file but command log)
> > > >
> > > >
> > > > > Thanks.
> > > > > S.
> > > > >
> > > > >
> > > > > Il giorno ven 29 gen 2021 alle ore 11:39 Paul B Mahol <
> > > one...@gmail.com>
> > > > > ha
> > > > > scritto:
> > > > >
> > > > > > On Fri, Jan 29, 2021 at 11:25 AM Marco Mircoli <
> > nuvolab...@gmail.com
> > > >
> > > > > > wrote:
> > > > > >
> > > > > > > Hello,
> > > > > > > anybody knows the reason why this command works.
> > > > > > >
> > > > > > > $ffmpegCmd = "ffmpeg -i $fileOriginale -af
> > > > > > >
> > > > > > >
> > > > > >
> > > > >
> > > >
> > >
> >
> adeclick,afftdn=nr=80:nf=-20:nt=w:om=o,highpass=f=70,loudnorm=I=-16:TP=-1.5:LRA=14
> > > > > > > $fileProcessato"; //Livella audio a R128
> > > > > > >
> > > > > > > and this doesn't work.
> > > > > > >
> > > > > > > $ffmpegCmd = "ffmpeg -i $fileOriginale -af
> > > > > > >
> > > > > > >
> > > > > >
> > > > >
> > > >
> > >
> >
> adeclick,deesser=i=1,afftdn=nr=80:nf=-20:nt=w:om=o,highpass=f=70,loudnorm=I=-16:TP=-1.5:LRA=14
> > > > > > > $fileProcessato"; //Livella audio a R128
> > > > > > > This creates an audio file of zero bytes, just the name.
> > > > > > >
> > > > > > > The difference is that the last I've added deesser=i=1
> > > > > > > Any ideas?
> > > > > > >
> > > > > >
> > > > > > Dunno, you have not provided much info.
> > > > > > Also both commands works just fine here.
> > > > > >
> > > > > >
> > > > > > > Thanks.
> > > > > > > S.
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > Il giorno gio 28 gen 2021 alle ore 10:15 Michael Koch <
> > > > > > > astroelectro...@t-online.de> ha scritto:
> > > > > > >
> > > > &

Re: [FFmpeg-user] VOICE POST PRODUCTION IN ONE LINE

[Paul B Mahol]

On Fri, Jan 29, 2021 at 12:52 PM Chris Miceli  wrote:

> If you wish to have output from your service, you would need to check the
> documentation of the library you are using to execute the shell command.
> They often have the ability to capture stdout and stderr which is what you
> need.
>
> Often, these libraries may interpret shell commands in different ways and
> so it may be parsing the ffmpeg command line (this is something to keep in
> mind, not necessarily what is happening).
>
> If you can run your command from the shell normally, there is likely
> something in your service code which is causing it.
>

The only thing that come to mind, is that version is old that it does not
have that specific filter.
Also note that those audio filters are not magic way to solve your problems
by putting them in filtergraph and forgetting about them next second.


> On Fri, Jan 29, 2021, 10:21 PM Paul B Mahol  wrote:
>
> > On Fri, Jan 29, 2021 at 12:18 PM Marco Mircoli 
> > wrote:
> >
> > > Thanks,
> > > is there a way to understand if ffmpeg has aborted the process?
> > > I'm using a Linux version for an online service.
> > > It would be nice to let ffmpeg write a log file when there is a problem
> > > (process aborted)
> > > How to do it?
> > >
> >
> > By providing full ffmpeg output. (Not output file but command log)
> >
> >
> > > Thanks.
> > > S.
> > >
> > >
> > > Il giorno ven 29 gen 2021 alle ore 11:39 Paul B Mahol <
> one...@gmail.com>
> > > ha
> > > scritto:
> > >
> > > > On Fri, Jan 29, 2021 at 11:25 AM Marco Mircoli  >
> > > > wrote:
> > > >
> > > > > Hello,
> > > > > anybody knows the reason why this command works.
> > > > >
> > > > > $ffmpegCmd = "ffmpeg -i $fileOriginale -af
> > > > >
> > > > >
> > > >
> > >
> >
> adeclick,afftdn=nr=80:nf=-20:nt=w:om=o,highpass=f=70,loudnorm=I=-16:TP=-1.5:LRA=14
> > > > > $fileProcessato"; //Livella audio a R128
> > > > >
> > > > > and this doesn't work.
> > > > >
> > > > > $ffmpegCmd = "ffmpeg -i $fileOriginale -af
> > > > >
> > > > >
> > > >
> > >
> >
> adeclick,deesser=i=1,afftdn=nr=80:nf=-20:nt=w:om=o,highpass=f=70,loudnorm=I=-16:TP=-1.5:LRA=14
> > > > > $fileProcessato"; //Livella audio a R128
> > > > > This creates an audio file of zero bytes, just the name.
> > > > >
> > > > > The difference is that the last I've added deesser=i=1
> > > > > Any ideas?
> > > > >
> > > >
> > > > Dunno, you have not provided much info.
> > > > Also both commands works just fine here.
> > > >
> > > >
> > > > > Thanks.
> > > > > S.
> > > > >
> > > > >
> > > > >
> > > > > Il giorno gio 28 gen 2021 alle ore 10:15 Michael Koch <
> > > > > astroelectro...@t-online.de> ha scritto:
> > > > >
> > > > > > Am 28.01.2021 um 09:44 schrieb Marco Mircoli:
> > > > > > > Goodmorning everyone.
> > > > > > > I'm new to ffmpeg and was wondering if anyone from the group
> > could
> > > > help
> > > > > > me
> > > > > > > set up a command line.
> > > > > > > My goal is to improve voice audio recordings.
> > > > > > > I would like to be able to do this starting from a command
> line I
> > > > have
> > > > > in
> > > > > > > my script, modifying it to add:
> > > > > > >
> > > > > > > - 70HZ high pass filter
> > > > > > > - expander / noise gate
> > > > > > > - equalizer
> > > > > > > - compressor
> > > > > > > - exciter
> > > > > > > - de-esser
> > > > > > > - limiter
> > > > > > > - loudnorm to r128 (there is already)
> > > > > > >
> > > > > > > It would be nice to have all this just in one line, is it
> > possible?
> > > > > > > This is the string where I'm starting from.
> > > > > > >
> > > > > > > $ shell = shell_exec ("$ ffmpeg_b -i $ audio_file_full_path
> -map
> > 0:

Re: [FFmpeg-user] minterpolate ...for Paul

[Paul B Mahol]

On Fri, Jan 29, 2021 at 2:01 AM pdr0  wrote:

> Mark Filipak (ffmpeg) wrote
> > I've never heard of "optical flow errors". What could they be? (Got any
> > links to
> > explanations?)
>
> The artifacts in your video are optical flow errors :)
>
> If you've ever used it - you'd recognize these artifacts. There are very
> common
>
>
>
> There are about a dozen prototypical "fail" categories or common errors
> that
> plague all types of optical flow
>
> These are errors either of motion vectors, or object flow (object
> boundaries
> or "masks"), occlusion errors.
>
> Internet is full of examples, explanations. The topic is rather large, just
> search google, there is lots of info. If you have a specific question then
> ask.
>
> Sometimes you get clean interpolated frame results;  but sometimes there
> are
> massive distracting errors. It varies by situation and sources.
>
> Your example has one of the common categories of "fail" where there are
> repeating patterns and textures. It falls under the "Picket Fence" fail . A
> prototypical tracking or dolly shot by a picket fence, or brick wall will
> come up with interpolation errors
>
> The peripheral edges error are common because there is less data beyond the
> periphery of the frame, for n-1, n+1 and the motion vectors are less
> accurate compared to the center of the frame
>
> Another common one is when objects pass over another. The flow masks aren't
> perfect and you end up with blobby edge artifacts around objects
>
>
>
>
>
>
>
> >>...For artifacts around frame edges, letterbox edges usually some form
> >> of padding is used. I don't think ffmpeg minterpolate has those.
> >
> > I've done that. The result was just okay. The slight riffling on the
> frame
> > boundaries during camera
> > panning isn't all that objectionable to me. It occurs to me that
> > minterpolute could queue frames and
> > look 'forward' to later frames in order to resolve boundary macroblock
> > artifacts -- afterall, it has
> > the motion vectors, eh?
>
> Some algorithms can use N-3, N-2, N-1, N, N+1, N+2, N+3, I don't think
> minterpolate can.  More is not always better. Often you get more
> contamination with a larger "window"
>
> Sometimes just changing the blocksize can produce better (or worse)
> results.
> The problem is ffmpeg minterpolate is s slow, and you have no usable
> preview. Some of the other methods mentioned earlier do have previews -  so
> you can tweak settings, preview, readjust etc
>
>
Why you ignore fact that libavfilter also allows usable preview and
readjust of parameters.

Amount of false information propagation here is astounding.


>
>
>
>
>
>
>
>
> --
> Sent from: http://ffmpeg-users.933282.n4.nabble.com/
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Re: [FFmpeg-user] VOICE POST PRODUCTION IN ONE LINE

[Paul B Mahol]

On Fri, Jan 29, 2021 at 12:18 PM Marco Mircoli  wrote:

> Thanks,
> is there a way to understand if ffmpeg has aborted the process?
> I'm using a Linux version for an online service.
> It would be nice to let ffmpeg write a log file when there is a problem
> (process aborted)
> How to do it?
>

By providing full ffmpeg output. (Not output file but command log)


> Thanks.
> S.
>
>
> Il giorno ven 29 gen 2021 alle ore 11:39 Paul B Mahol 
> ha
> scritto:
>
> > On Fri, Jan 29, 2021 at 11:25 AM Marco Mircoli 
> > wrote:
> >
> > > Hello,
> > > anybody knows the reason why this command works.
> > >
> > > $ffmpegCmd = "ffmpeg -i $fileOriginale -af
> > >
> > >
> >
> adeclick,afftdn=nr=80:nf=-20:nt=w:om=o,highpass=f=70,loudnorm=I=-16:TP=-1.5:LRA=14
> > > $fileProcessato"; //Livella audio a R128
> > >
> > > and this doesn't work.
> > >
> > > $ffmpegCmd = "ffmpeg -i $fileOriginale -af
> > >
> > >
> >
> adeclick,deesser=i=1,afftdn=nr=80:nf=-20:nt=w:om=o,highpass=f=70,loudnorm=I=-16:TP=-1.5:LRA=14
> > > $fileProcessato"; //Livella audio a R128
> > > This creates an audio file of zero bytes, just the name.
> > >
> > > The difference is that the last I've added deesser=i=1
> > > Any ideas?
> > >
> >
> > Dunno, you have not provided much info.
> > Also both commands works just fine here.
> >
> >
> > > Thanks.
> > > S.
> > >
> > >
> > >
> > > Il giorno gio 28 gen 2021 alle ore 10:15 Michael Koch <
> > > astroelectro...@t-online.de> ha scritto:
> > >
> > > > Am 28.01.2021 um 09:44 schrieb Marco Mircoli:
> > > > > Goodmorning everyone.
> > > > > I'm new to ffmpeg and was wondering if anyone from the group could
> > help
> > > > me
> > > > > set up a command line.
> > > > > My goal is to improve voice audio recordings.
> > > > > I would like to be able to do this starting from a command line I
> > have
> > > in
> > > > > my script, modifying it to add:
> > > > >
> > > > > - 70HZ high pass filter
> > > > > - expander / noise gate
> > > > > - equalizer
> > > > > - compressor
> > > > > - exciter
> > > > > - de-esser
> > > > > - limiter
> > > > > - loudnorm to r128 (there is already)
> > > > >
> > > > > It would be nice to have all this just in one line, is it possible?
> > > > > This is the string where I'm starting from.
> > > > >
> > > > > $ shell = shell_exec ("$ ffmpeg_b -i $ audio_file_full_path -map 0:
> > a:
> > > 0
> > > > > -af loudnorm -b: a 96k $ audio_output_mp3 2> & 1");
> > > >
> > > > yes, I think it's possible to do all this in one command line. All
> > these
> > > > features must be added at the beginning of the audio filter chain,
> > > > that's between "-af" and "loudnorm". For example:
> > > >
> > > > $ shell = shell_exec ("$ ffmpeg_b -i $ audio_file_full_path -map 0:
> a:
> > 0
> > > > -af highpass=f=70,loudnorm -b: a 96k $ audio_output_mp3 2> & 1");
> > > >
> > > > Michael
> > > > ___
> > > > ffmpeg-user mailing list
> > > > ffmpeg-user@ffmpeg.org
> > > > https://ffmpeg.org/mailman/listinfo/ffmpeg-user
> > > >
> > > > To unsubscribe, visit link above, or email
> > > > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
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> > >
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Re: [FFmpeg-user] ffmpeg, amix, adelay out of sync issues

[Paul B Mahol]

On Fri, Jan 22, 2021 at 11:20 PM Randy Johnson via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

> Hello,
>
> I have a series of mp3 files with a start time offset.
> I am trying to merge them together with adelay and amix.
> The following codes run and creates the combined MP3 but when I listen
> to the final output file there are spots where people are talking over
> each other not in the sense they are being rude but the audio being out
> of sync.
> Here are the files with the start and stoptime offsets:
>
> d4a19a58-790a-45c1-b96e-819bc98cd408.webm | 1720 | 3762173 ab794f92-
> bdfb-4bb0-821d-ba80c8a9228a.webm | 48920 | 3328385 a7df9a51-0671-4930-
> a14b-e7d85a562577.webm | 76220 | 3763361 5d312878-d25d-40d7-9418-
> bed85e22efc6.webm | 3109640 | 3267473 1cf6127d-250a-4601-a219-
> b1b6bbd725f8.webm | 3360980 | 3636927 c3ddd462-e05a-425f-af58-
> f288f930f2cb.webm | 3689600 | 3760423
> Here is the command broke into lines for easier readability:
>
> ffmpeg -i d4a19a58-790a-45c1-b96e-819bc98cd408.mp3 -i ab794f92-bdfb-
> 4bb0-821d-ba80c8a9228a.mp3 -i a7df9a51-0671-4930-a14b-e7d85a562577.mp3 -
> i 1cf6127d-250a-4601-a219-b1b6bbd725f8.mp3 -i c3ddd462-e05a-425f-af58-
> f288f930f2cb.mp3 -filter_complex " [0]adelay=1720|1720[a0];
> [1]adelay=48920|48920[a1]; [2]adelay=76220|76220[a2];
> [3]adelay=3360980|3360980[a3]; [4]adelay=3689600|3689600[a4];
> [a0][a1][a2][a3][a4]amix=inputs=5 " out.mp3 -y
> Any idea what I am doing wrong?
>

If you want real help, provide input files.

Also you have not provided information like ffmpeg version.

Always use latest master git build version.


>
> Thanks!
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Re: [FFmpeg-user] VOICE POST PRODUCTION IN ONE LINE

[Paul B Mahol]

On Fri, Jan 29, 2021 at 11:25 AM Marco Mircoli  wrote:

> Hello,
> anybody knows the reason why this command works.
>
> $ffmpegCmd = "ffmpeg -i $fileOriginale -af
>
> adeclick,afftdn=nr=80:nf=-20:nt=w:om=o,highpass=f=70,loudnorm=I=-16:TP=-1.5:LRA=14
> $fileProcessato"; //Livella audio a R128
>
> and this doesn't work.
>
> $ffmpegCmd = "ffmpeg -i $fileOriginale -af
>
> adeclick,deesser=i=1,afftdn=nr=80:nf=-20:nt=w:om=o,highpass=f=70,loudnorm=I=-16:TP=-1.5:LRA=14
> $fileProcessato"; //Livella audio a R128
> This creates an audio file of zero bytes, just the name.
>
> The difference is that the last I've added deesser=i=1
> Any ideas?
>

Dunno, you have not provided much info.
Also both commands works just fine here.


> Thanks.
> S.
>
>
>
> Il giorno gio 28 gen 2021 alle ore 10:15 Michael Koch <
> astroelectro...@t-online.de> ha scritto:
>
> > Am 28.01.2021 um 09:44 schrieb Marco Mircoli:
> > > Goodmorning everyone.
> > > I'm new to ffmpeg and was wondering if anyone from the group could help
> > me
> > > set up a command line.
> > > My goal is to improve voice audio recordings.
> > > I would like to be able to do this starting from a command line I have
> in
> > > my script, modifying it to add:
> > >
> > > - 70HZ high pass filter
> > > - expander / noise gate
> > > - equalizer
> > > - compressor
> > > - exciter
> > > - de-esser
> > > - limiter
> > > - loudnorm to r128 (there is already)
> > >
> > > It would be nice to have all this just in one line, is it possible?
> > > This is the string where I'm starting from.
> > >
> > > $ shell = shell_exec ("$ ffmpeg_b -i $ audio_file_full_path -map 0: a:
> 0
> > > -af loudnorm -b: a 96k $ audio_output_mp3 2> & 1");
> >
> > yes, I think it's possible to do all this in one command line. All these
> > features must be added at the beginning of the audio filter chain,
> > that's between "-af" and "loudnorm". For example:
> >
> > $ shell = shell_exec ("$ ffmpeg_b -i $ audio_file_full_path -map 0: a: 0
> > -af highpass=f=70,loudnorm -b: a 96k $ audio_output_mp3 2> & 1");
> >
> > Michael
> > ___
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> >
> > To unsubscribe, visit link above, or email
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Re: [FFmpeg-user] minterpolate performance & alternative

[Paul B Mahol]

On Thu, Jan 28, 2021 at 10:23 PM Mark Filipak (ffmpeg) 
wrote:

> Synopsis:
>
> I seek to use minterpolate to take advantage of its superior output. I
> present some performance
> issues followed by an alternative filter_complex. So, this presentation
> necessarily addresses 2
> subjects.
>
> Problem:
>
> I'm currently transcoding a 2:43:05 1920x1080, 24FPS progressive video to
> 60FPS via minterpolate
> filter. Apparently, the transcode will take a little more than 3 days.
>
> Hardware:
>
> There are 4 CPU cores (with 2 threads, each) that run at 3.6 GHz. There is
> also an NVIDIA GTX 980M
> GPU having 1536 CUDA cores with a driver that implements the Optimus,
> CUDA-as-coprocessors architecture.
>
> Performance:
>
> During the transcode, ffmpeg is consuming only between 10% & 20% of the
> CPU. It appears to be
> single-threaded, and it appears to not be using Optimus at all.
>
> Is there a way to coax minterpolate to expand its hardware usage?
>
> Alternative filter_complex:
>
> minterpolate converts 24FPS to 60FPS by interpolating every frame via
> motion vectors to produce a 60
> picture/second stream in a 60FPS transport. It does a truly amazing job,
> but without expanded
> hardware usage, it takes too long to do it.
>
> A viable alternative is to 55 telecine the source (which simply duplicates
> the n%5!=2 frames) while
> interpolating solely the n%5==2 frames. That should take much less time
> and would produce a 24
> picture/second stream in a 60FPS transport -- totally acceptable.
>
> The problem is that motion vector interpolation requires that minterpolate
> be 'split' out and run in
> parallel with the main path in the filter_complex so that the interpolated
> frames can be plucked out
> (n%5==2) and interleaved at the end of the filter_complex. That doesn't
> make much sense because it
> doesn't decrease processing (or processing time) and, if the fully
> motion-interpolated stream is
> produced anyway, then output it directly instead of interleaving. What's
> needed is an interpolation
> alternative to minterpolate.
>
> Alternative Interpolation:
>
> 55 telecine with no interpolation or smoothing works well even though the
> n%5==2 frames are combed
> but decombing is desired. The problem with that is: I can't find a
> deinterlace filter that does
> pixel interpolation without reintroducing some telecine judder. The issue
> involves spacial alignment
> of the odd & even lines in the existing filters.
>
> Some existing filters align the decombed lines with the input's top field,
> some align the decombed
> lines with the input's bottom field. What's desired is a filter that
> aligns the decombed lines with
> the spacial mean. I suggest that the Sobel might be appropriate for the
> decombing (or at least, that
> the Sobel can be employed to visualize what's desired).
>
> Sobel of line y:   __/\_/\_ (edges)
> Sobel of line y+1: __/\_/\_
> Desired output:
>   line y:   /\_/\___ (aligned to mean)
>   line y+1: /\_/\___ (aligned to mean)
> I could find this:
>   line y:   __/\_/\_
>   line y+1: __/\_/\_ (aligned to top line
> edges)
> and I could find this:
>   line y:   __/\_/\_ (aligned to bottom
> line edges)
>   line y+1: __/\_/\_
>
>
Sorry, but I can not decipher above stuff. Does anybody else can?


> Does such a decomb filter exist? I've tested every deinterlacing filter,
> and I've failed to find one.
>
> Thank you for reading all this. I'm sorry it's so long.
>
> Regards,
> Mark.
> --
> Someone's sneaking in and turning up the range so that my food burns.
> I'm sure of it.
> And the older I get, the more sure of it I become.
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Re: [FFmpeg-user] minterpolate problem

[Paul B Mahol]

On Thu, Jan 28, 2021 at 7:04 AM Mark Filipak (ffmpeg) 
wrote:

> On 01/27/2021 05:47 PM, Carl Eugen Hoyos wrote:
> > Am Mi., 27. Jan. 2021 um 02:03 Uhr schrieb Mark Filipak (ffmpeg)
> > :
> >
> >> ffmpeg -i SOURCE -map 0 -filter_complex "split[1][2], \
> >> [1]telecine=pattern=55, select=not(eq(mod(n\,5)\,2))[3], \
> >>
> [2]minterpolate=fps=6/1001:mi_mode=mci=mc_mode=obmc:scd=fdiff:scd_threshold=10,
> \
> >> select=eq(mod(n\,5)\,2)[4], \
> >> [3][4]interleave" \
> >> -codec:v libx265 -x265-params "crf=16:qcomp=0.60" -codec:a copy
> -codec:s copy TARGET
> >
> > Since others may read this:
> > The command line makes little sense, you should either use
> > telecine or minterpolate.
>
> Carl Eugen, you surprise me. I'm sure you see what I'm trying to do, so
> why the angst?
>
> First, let me admit that since I started this thread (following weeks of
> experimentation) I decided
> to simply try this: 'minterpolate=fps=6/1001' to see what it would do.
> I was amazed.
>
> If minterpolate is Paul's work, then Paul is a f'ing genius.
>
> I've since spent a couple of days playing with minterpolate and have found
> that this:
>
> 'minterpolate=fps=6/1001:mi_mode=mci:mc_mode=obmc:scd=fdiff:scd_threshold=10:vsbmc=1:search_param=20'
> is astounding.
>
> Nonetheless, I'd still like to generate a 60 FPS transcode that is
> visually equivalent (as closely
> as possible) to 24 pictures/second by replacing the combed frame (i.e.
> n%5==2) by an interpolated
> frame -- hence the filter graph you loathe.
>
> For that n%5==2 frame, here's what I sought (and did not find):
>
> Sobel of line y:   __/\_/\_
> Sobel of line y+1: __/\_/\_
> Desired output:
>   line y:   /\_/\___
>   line y+1: /\_/\___
> I could find this:
>   line y:   __/\_/\_
>   line y+1: __/\_/\_ (aligned to top line)
> and I could find this:
>   line y:   __/\_/\_ (aligned to bottom
> line)
>   line y+1: __/\_/\_
> but I could not find this:
>   line y:   /\_/\___ (aligned to mean)
>   line y+1: /\_/\___ (aligned to mean)
>
>
Please cite reference/paper that explains this technique in more detail.



> One reason I might -- MIGHT -- prefer telecine55 is that the original
> frames are recoverable from
> the transcodes -- but admittedly, that's not much of a reason, is it?
> Another reason is that the
> transcode is much, much quicker than 'minterpolate=fps=6/1001' and
> somewhat smaller.
>
> There you have it, Carl Eugen. Please stop trying to make the world think
> I'm feeble minded.  :-)
> Okay?
>
> Regards,
> Mark.
>
> --
> Someone's sneaking in and turning up the range so that my food burns.
> I'm sure of it.
> And the older I get, the more sure of it I become.
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Re: [FFmpeg-user] minterpolate problem

[Paul B Mahol]

On Wed, Jan 27, 2021 at 6:03 AM Mark Filipak (ffmpeg) 
wrote:

> I ask this because I think it can only be answered by looking at the
> source code. I'm not competent
> to do that.
>

Your thinking is invalid.


>
> Is 'mi_mode=dup' or 'mi_mode=blend' or 'mi_mode=mci' the default?
>


ffmpeg -h filter=minterpolate


>
>
> On 01/26/2021 09:49 PM, pdr0 wrote:
> > Jim DeLaHunt-2 wrote
> >> Perhaps the character between 'mci' and 'mc_mode' should be ':' instead
> >> of '='?
> >
> > That works for me
> >
> > -vf
> >
> minterpolate=fps=6/1001:mi_mode=mci:mc_mode=obmc:scd=fdiff:scd_threshold=10
> >
> > Each one is a separate option and argument
> >
> >
> https://github.com/FFmpeg/FFmpeg/blob/master/libavfilter/vf_minterpolate.c
>
> Thank you. It works for me, too. I don't know why it didn't work the 1st
> time (except maybe because
> I have so many balls in the air).
>
> So, unlike what is shown in the documentation, 'mc_mode' is _not_ an
> argument of 'mci'.
>
> Apparently, 1, 'mi_mode=mci' is a valid argument and, 2, that 'mc_mode'
> (and perhaps 'me_mode' &
> 'me') are valid solely if mi_mode=mci but are otherwise not arguments of
> 'mi_mode=mci' but are
> direct options of 'minterpolate'. That kind of scope is probably what the
> author intended when
> he/she indented them below 'mci', but the indenting is very misleading.
>
> I will try all combinations of minterpolate to resolve all the options and
> issues and return with a
> new beautified version of the minterpolate documentation. Perhaps that
> will save people in the
> future because minterpolate is an important and amazing function.
>
> --
> Someone's sneaking in and turning up the range so that my food burns.
> I'm sure of it.
> And the older I get, the more sure of it I become.
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Re: [FFmpeg-user] Next Release

[Paul B Mahol]

On Tue, Jan 26, 2021 at 8:44 PM Phil Rhodes via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

>  On Tuesday, 26 January 2021, 19:10:38 GMT, Chris Angelico <
> ros...@gmail.com> wrote:
>
> > Do you realise how toxic you make the FFMPEG community look?
>
> To answer your last point first, either they don't know how toxic it
> looks, or they don't care. It's been like this for so long that if it's an
> elaborate trolling attempt it's a very, very determined one. Either way
> it's horrible and I regret that it exists, but nobody in authority seems
> willing to do anything about it.
>
> As regards the issue of release versions, the way I usually do it is to
> download a statically-built executable. As a windows person I'd invariably
> do this anyway; I have built ffmpeg on windows but it's a real hurt in the
> hiney. There are usually executables available which presumably represent
> "current git head" as it existed at some point in time. It's anecdotal, but
> I've never had a problem with these. You can update at will, or keep your
> known-good setup together for the price of simply including that executable
> file.
>
> There are, as I understand it, a number of potential legal problems with
> distributing that, since it is quite difficult to obtain a set of source
> files that are known to be symmetrical with the executable you have. It's
> even more difficult to prove that any particular set of source files are
> symmetrical with any particular executable. Still, it seems to be more or
> less tolerated on the basis that as a purely practical matter, anyone who
> wants the source to ffmpeg would download it anyway.
>
>
Yes, if you want to use ffmpeg you need to pay for each patent it
implements.



> P
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Re: [FFmpeg-user] Next Release

[Paul B Mahol]

On Tue, Jan 26, 2021 at 7:12 PM Carl Zwanzig  wrote:

> On 1/26/2021 10:03 AM, Reindl Harald wrote:
> > Am 26.01.21 um 18:59 schrieb Paul B Mahol:
> >> This is invalid
>
> I'm starting to think that when Paul says "invalid" he means "I don't
> agree
> with you".
>
> > it is *not* invalid
> >
> > nobody on this planet can assure that a random git-snapshot is error
> free
> > and everyone who pretends that is either stupid or lying
>
> Have to agree, pretty much _every_other_ opensource project* manages x.y
> or
> x.y.z releases and _doesn't_ support (whatever that even means) their git
> head or development branches.
>

That is invalid statements from you. just add it to pile of others.


>
> *(I'm sure there are exceptions to that, but few and far between.)
>
> Later,
>
> z!
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Re: [FFmpeg-user] Next Release

[Paul B Mahol]

On Tue, Jan 26, 2021 at 7:13 PM Reindl Harald 
wrote:

>
>
> Am 26.01.21 um 19:09 schrieb Paul B Mahol:
> > On Tue, Jan 26, 2021 at 7:04 PM Reindl Harald 
> > wrote:
> >
> >>
> >>
> >> Am 26.01.21 um 18:59 schrieb Paul B Mahol:
> >>> On Tue, Jan 26, 2021 at 6:50 PM Reindl Harald 
> >>> wrote:
> >>>
> >>>>
> >>>>
> >>>> Am 26.01.21 um 18:09 schrieb Carl Eugen Hoyos:
> >>>>> Am Di., 26. Jan. 2021 um 10:33 Uhr schrieb Armin Hasitzka <
> >>>> ar...@grabyo.com>:
> >>>>>
> >>>>>> Given that FFmpeg has been released on an almost
> >>>>>> monthly basis before that
> >>>>>
> >>>>> Nearly one year passed between 4.2 and 4.3.
> >>>>
> >>>> you heard about point releases?
> >>>>
> >>>>> We recommend that all users download current
> >>>>> FFmpeg git head as we lack the manpower to
> >>>>> support other versions
> >>>>
> >>>> which manpower does it take to publish point releases like everyone
> but
> >>>> you does?
> >>>>
> >>>> nobody right in his brain daily builds software from git and no,
> current
> >>>> git is not always the stuff with less errors than anything else, you
> >>>> can't asure that
> >>>>
> >>>
> >>> This is invalid
> >>
> >> it is *not* invalid
> >>
> >> nobody on this planet can assure that a random git-snapshot is error
> >> free and everyone who pretends that is either stupid or lying
> >>
> >
> > It for sure have less bugs and more features than any other release
>
> stop takling bullshit!
>
> you can't assure mistakes and regressions don't happen in random commits
>

No, we have FATE. Everything is cached. Watch your tone.

- period
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Re: [FFmpeg-user] Next Release

[Paul B Mahol]

On Tue, Jan 26, 2021 at 7:09 PM Armin Hasitzka  wrote:

> > Nearly one year passed between 4.2 and 4.3.
> >
> > We recommend that all users download current
> > FFmpeg git head as we lack the manpower to
> > support other versions.
>
> Fair enough, thanks for that insight :) Out of curiosity -- are there
> any special (stability) checks you run for releasing versions (that
> you don't run on every commit to `master`) or are releases just dates
> when you fire up all build pipelines + update the website? Just trying
> to understand whether there's _any_ upside to waiting for a release
> instead of just grabbing `master` today.
>

For release versions, one just make sure that FATE report is not yellow.


>
> Armin
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Re: [FFmpeg-user] Next Release

[Paul B Mahol]

On Tue, Jan 26, 2021 at 7:04 PM Reindl Harald 
wrote:

>
>
> Am 26.01.21 um 18:59 schrieb Paul B Mahol:
> > On Tue, Jan 26, 2021 at 6:50 PM Reindl Harald 
> > wrote:
> >
> >>
> >>
> >> Am 26.01.21 um 18:09 schrieb Carl Eugen Hoyos:
> >>> Am Di., 26. Jan. 2021 um 10:33 Uhr schrieb Armin Hasitzka <
> >> ar...@grabyo.com>:
> >>>
> >>>> Given that FFmpeg has been released on an almost
> >>>> monthly basis before that
> >>>
> >>> Nearly one year passed between 4.2 and 4.3.
> >>
> >> you heard about point releases?
> >>
> >>> We recommend that all users download current
> >>> FFmpeg git head as we lack the manpower to
> >>> support other versions
> >>
> >> which manpower does it take to publish point releases like everyone but
> >> you does?
> >>
> >> nobody right in his brain daily builds software from git and no, current
> >> git is not always the stuff with less errors than anything else, you
> >> can't asure that
> >>
> >
> > This is invalid
>
> it is *not* invalid
>
> nobody on this planet can assure that a random git-snapshot is error
> free and everyone who pretends that is either stupid or lying
>

It for sure have less bugs and more features than any other release.


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Re: [FFmpeg-user] Next Release

[Paul B Mahol]

On Tue, Jan 26, 2021 at 6:50 PM Reindl Harald 
wrote:

>
>
> Am 26.01.21 um 18:09 schrieb Carl Eugen Hoyos:
> > Am Di., 26. Jan. 2021 um 10:33 Uhr schrieb Armin Hasitzka <
> ar...@grabyo.com>:
> >
> >> Given that FFmpeg has been released on an almost
> >> monthly basis before that
> >
> > Nearly one year passed between 4.2 and 4.3.
>
> you heard about point releases?
>
> > We recommend that all users download current
> > FFmpeg git head as we lack the manpower to
> > support other versions
>
> which manpower does it take to publish point releases like everyone but
> you does?
>
> nobody right in his brain daily builds software from git and no, current
> git is not always the stuff with less errors than anything else, you
> can't asure that
>

This is invalid.


>
> how do you imagine that to scale in case not only ffmpeg but every
> single piece of software has that attitude?
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Re: [FFmpeg-user] Troublemaker in bugtracker

[Paul B Mahol]

On Tue, Jan 26, 2021 at 1:18 AM Phil Rhodes via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

>  On Monday, 25 January 2021, 22:34:35 GMT, Mark Filipak (ffmpeg) <
> markfili...@bog.us> wrote:
>
> > Your comments, Leo, are quite correct. However, there is no F*ing
> Manual. There is only man-style
> > documentation that supplies usage reminders to those who are already
> familiar with ffmpeg and its > usage, but that doesn't help newcomers. I
> wish the reality was different, and I've volunteered to > help with
> documentation, but the developers have rebuffed me and everyone else who
> has similarly > volunteered.
> Very much so. I may be one of the volunteers Mark is referring to.
> In the end it's a very long term problem that exists with a lot of open
> source projects and I don't see a solution to it. When you have people who
> are determined to treat others with contempt and hostility as a matter of
> course, who reject even the most well-meaning offers of help with the most
> pressing issues, who almost seem to make a science out of deliberate,
> studied unpleasantness, something strange is going on.
> More than annoying, it's weird. I've never once in my life encountered
> anything like this. It's completely unnecessary, completely
> self-destructive, and appears to gain nothing for anyone.
>

Yes, we need to use only corporate code.


> P
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Re: [FFmpeg-user] overlaying animated PNG, doesn't loop

[Paul B Mahol]

On Mon, Jan 25, 2021 at 3:45 PM Phil Rhodes via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

>  On Monday, 25 January 2021, 14:41:36 GMT, Paul B Mahol 
> wrote:
>
> > You are very ignorant of mentioned demuxer options in documentation.
> You've got to be kidding.
> Paul, you are coming off as unbelievably unpleasant. Again, I'm trying to
> help you. Please, please, moderate your tone.
>

You should be immediately banned from this list.


> P
>
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Re: [FFmpeg-user] overlaying animated PNG, doesn't loop

[Paul B Mahol]

On Mon, Jan 25, 2021 at 3:12 PM Dave Thorn  wrote:

> On Mon, Jan 25, 2021 at 11:29:39AM +0100, Paul B Mahol wrote:
> > On Mon, Jan 25, 2021 at 10:21 AM Dave Thorn via ffmpeg-user <
> ffmpeg-user@ffmpeg.org> wrote:
> []
> > > Anyone know anything about this, please, or will I have to avoid
> > > APNGs?
> > >
> > >
> > Next time read documentation before you write something:
> >
> > http://ffmpeg.org/ffmpeg-formats.html#apng
>
> I hate to be the thickie, but does that explain why it doesn't loop?
> The acTL is before the fcTL so that means it is "transmitted as
> extradata"?  What does that mean to me-the-user?
>

You are very ignorant of mentioned demuxer options in documentation.


>
> Cheers,
>
> --
> dave thorn
>
> DBT; GBP.
>
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Re: [FFmpeg-user] overlaying animated PNG, doesn't loop

[Paul B Mahol]

On Mon, Jan 25, 2021 at 10:21 AM Dave Thorn via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

>
> On Mon, Jan 11, 2021 at 08:08:03AM +, Dave Thorn via ffmpeg-user wrote:
> > Yeah, sorry.  Stuck on Windows at the moment and I was anticipating
> > difficulty updating, but it was fine.
> >
> > Same behaviour in this version:
>
> [snip]
>
> Anyone know anything about this, please, or will I have to avoid
> APNGs?
>
>
Next time read documentation before you write something:

http://ffmpeg.org/ffmpeg-formats.html#apng


> Thanks,
>
>
> --
> dave thorn
>
> Bad to the bone.
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Re: [FFmpeg-user] 7.1 wave filter complex amerge issue

[Paul B Mahol]

On Sun, Jan 24, 2021 at 11:39 AM Bouke  wrote:

>
>
> > On 24 Jan 2021, at 11:29, Paul B Mahol  wrote:
> >
> > On Sun, Jan 24, 2021 at 11:26 AM Bouke  bo...@editb.nl>> wrote:
> >
> >> Trying to remap channels in a 7.1 Wave file, but that gives me a strange
> >> error.
> >>
> >> (I’ve reduced my lines to the most simple to reproduce.)
> >>
> >> When I use the same input file, but do a 5.1 remap it works as expected:
> >> /Applications/ffmpeg -i /Volumes/Data/LC_testfiles/8chan.wav
> >> -filter_complex
> >>
> "[0:0][0:0][0:0][0:0][0:0][0:0]amerge=inputs=6,pan=5.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|[mix]"
> >> -map "[mix]" -f null -
> >>
> >> BUT, with 7.1 it fails
> >> /Applications/ffmpeg -i /Volumes/Data/LC_testfiles/8chan.wav
> >> -filter_complex
> >>
> "[0:0][0:0][0:0][0:0][0:0][0:0][0:0][0:0]amerge=inputs=8,pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|c6=c6|c7=c7|[mix]"
> >> -map "[mix]" -f null -
> >>
> >> Is it me, or is this a bug?
> >>
> >> Bouke
> >>
> >>
> >
> > I thought you gonna amerge mono channels into 7.1, but you use already
> 7.1
> > input.
> >
>
> Yes, I know. This is the simplest way of showing that there is something
> strange going on.
> The aim is to do a channel remap to get a correct channel layout for the
> R128 filter for eg
> So the mapping can be different, and could be something like:
> -filter_complex
> "[0:0][0:0][0:0][0:0][0:0][0:0][0:0][0:0]amerge=inputs=8,pan=7.1|c0=c0|c1=c2|c2=c1|c3=c4|c4=c5|c5=c6|c6=c7|c7=c3|[mix];[mix]ebur128=peak=true"
> -f null -
>

amerge filter merges channels, so if inputs have 8 channels each, output
will have 64 channels.
I think the limitation is there because channels layouts have support for
limited number of channels, and 64 channels are definitely not supported by
some filters because libswresample will not convert to different sample
format because of unknown layout of 64 channels input.


>
> Bouke
>
> >
> >> Failed to inject frame into filter network: Invalid argument
> >> Error while processing the decoded data for stream #0:0
> >> Conversion failed!
> >> ___
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Re: [FFmpeg-user] 7.1 wave filter complex amerge issue

[Paul B Mahol]

On Sun, Jan 24, 2021 at 11:26 AM Bouke  wrote:

> Trying to remap channels in a 7.1 Wave file, but that gives me a strange
> error.
>
> (I’ve reduced my lines to the most simple to reproduce.)
>
> When I use the same input file, but do a 5.1 remap it works as expected:
> /Applications/ffmpeg -i /Volumes/Data/LC_testfiles/8chan.wav
> -filter_complex
> "[0:0][0:0][0:0][0:0][0:0][0:0]amerge=inputs=6,pan=5.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|[mix]"
> -map "[mix]" -f null -
>
> BUT, with 7.1 it fails
> /Applications/ffmpeg -i /Volumes/Data/LC_testfiles/8chan.wav
> -filter_complex
> "[0:0][0:0][0:0][0:0][0:0][0:0][0:0][0:0]amerge=inputs=8,pan=7.1|c0=c0|c1=c1|c2=c2|c3=c3|c4=c4|c5=c5|c6=c6|c7=c7|[mix]"
> -map "[mix]" -f null -
>
> Is it me, or is this a bug?
>
> Bouke
>
> Output:
> ffmpeg version 4.3.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
> 2000-2020 the FFmpeg developers
>   built with Apple clang version 11.0.0 (clang-1100.0.33.17)
>   configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
> --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
> --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
> --enable-libfreetype --enable-libgsm --enable-libmodplug
> --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
> --enable-libopus --enable-librubberband --enable-libshine
> --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
> --enable-libtwolame --enable-libvidstab --enable-libvmaf
> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
> --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxavs --enable-libxvid --enable-libzimg --enable-libzmq
> --enable-libzvbi --enable-version3 --pkg-config-flags=--static
> --disable-ffplay
>   libavutil  56. 51.100 / 56. 51.100
>   libavcodec 58. 91.100 / 58. 91.100
>   libavformat58. 45.100 / 58. 45.100
>   libavdevice58. 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
>   libpostproc55.  7.100 / 55.  7.100
> Input #0, wav, from '/Volumes/Data/LC_testfiles/8chan.wav':
>   Metadata:
> encoded_by  : Pro Tools
> originator_reference: aaizN!wd5kQk
> date: 2021-01-22
> creation_time   : 16:15:18
> time_reference  : 0
> umid:
> 0x060A2B340101010501010F101300A2B90D6FE7648000E87D71677E54073C
>   Duration: 00:07:51.07, bitrate: 9264 kb/s
> Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1,
> s32 (24 bit), 9216 kb/s
> Stream mapping:
>   Stream #0:0 (pcm_s24le) -> amerge:in0
>   Stream #0:0 (pcm_s24le) -> amerge:in1
>   Stream #0:0 (pcm_s24le) -> amerge:in2
>   Stream #0:0 (pcm_s24le) -> amerge:in3
>   Stream #0:0 (pcm_s24le) -> amerge:in4
>   Stream #0:0 (pcm_s24le) -> amerge:in5
>   Stream #0:0 (pcm_s24le) -> amerge:in6
>   Stream #0:0 (pcm_s24le) -> amerge:in7
>   pan -> Stream #0:0 (pcm_s16le)
> Press [q] to stop, [?] for help
> [Parsed_amerge_0 @ 0x7f8654504240] No channel layout for input 1
> [Parsed_amerge_0 @ 0x7f8654504240] Input channel layouts overlap: output
> layout will be determined by the number of distinct input channels
> [auto_resampler_0 @ 0x7f8654508300] [SWR @ 0x7f8655811000] Value
> 2147483647.00 for parameter 'ich' out of range [0 - 64]
> [auto_resampler_0 @ 0x7f8654508300] [SWR @ 0x7f8655811000] Value
> 2147483647.00 for parameter 'och' out of range [0 - 64]
> [auto_resampler_0 @ 0x7f8654508300] [SWR @ 0x7f8655811000] Input channel
> count and layout are unset
> [auto_resampler_0 @ 0x7f8654508300] Failed to configure output pad on
> auto_resampler_0
> Error reinitializing filters!
>

I thought you gonna amerge mono channels into 7.1, but you use already 7.1
input.


> Failed to inject frame into filter network: Invalid argument
> Error while processing the decoded data for stream #0:0
> Conversion failed!
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Re: [FFmpeg-user] Troublemaker in bugtracker

[Paul B Mahol]

People should just ignore Carl Zwanzing and Phil Rhodes entries after all.

They will eventually leave when they see they can not influence users.

On Sun, Jan 24, 2021 at 4:23 AM Carl Zwanzig  wrote:

> Proof or it didn't happen? Did you just call Phil a liar? To what purpose?
>
> On 1/23/2021 6:16 PM, Bouke wrote:
> > What exactly are you trying to accomplish other than making a fool of
> yourself?
>
> One might ask the same thing. Or one could stop arguing about it,
> acknowledge that some posters here are brusque and unhelpful, and start
> trying to give useful answers to people's questions.
>
> (I don't think I've been threatened by Carl, but have by Paul- on
> 20-Dec-2020.)
>
> z!
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Re: [FFmpeg-user] colorchannelmixer: perceptual gamma

[Paul B Mahol]

On Sat, Jan 23, 2021 at 6:32 PM Simon Taylor 
wrote:

> Question: Is there a way to make ffmpeg's colorchannelmixer use perceptual
> gamma?
>

Added pl option to colorchannelmixer. It is designed to preserve overall
rgb lightness.
Its behavior is similar but not exact with gimp. So with your input using
also pl=1 option
will remove blue tint.
You need latest ffmpeg build with that commit included.


>
> Background: I'm trying to salvage source H.264 video from a capture device
> (8mm film scanner) which seems to have an extreme saturation issue that
> often leaves the red and green channels saturated with minimal blue,
> leaving a bright yellow stain on that area of the frame.
> https://i.imgur.com/6Qh6PRk.png
>
> Editing a single frame capture of the video in GIMP, I see that if I use
> the Colours -> Components -> Channel Mixer tool and supplement the blue
> channel (1.0) with the green channel (1.0) https://i.imgur.com/Q0qqgSP.png
> the result is desaturated but usable https://i.imgur.com/t19QU4U.png
>
> I've attempted to replicate this using ffmpeg's colorchannelmixer with the
> same values (output blue = 1.0*blue + 1.0*green)
>
> Code: https://pastebin.com/raw/czS76ZrN
>
> but the result has a blue tint. https://i.imgur.com/TsnBKb2.png
>
> Looking at the RGB values, ffmpeg's behaviour is technically correct (the
> processed B channel is an exact sum of the source B and G channels), but
> GIMP's behaviour is what I need.
>
> GIMP's behaviour is described in
>
> https://graphicdesign.stackexchange.com/questions/136750/gimp-channel-mixer-does-not-work-as-expected
> and gamma correction is suggested as the likely difference in how GIMP
> reaches its result.
>
> Thanks,
>
> Simon
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Re: [FFmpeg-user] colorchannelmixer: perceptual gamma

[Paul B Mahol]

On Sat, Jan 23, 2021 at 8:29 PM Phil Rhodes via ffmpeg-user <
ffmpeg-user@ffmpeg.org> wrote:

>  > I see that if I use
> > the Colours -> Components -> Channel Mixer tool and supplement the
> blue> channel (1.0) with the green channel (1.0)
> https://i.imgur.com/Q0qqgSP.png> the result is desaturated but usable
> https://i.imgur.com/t19QU4U.png
> I'd be a little cautious about this; have you tried it on other subjects?
> If it's just this clip then fine - I suspect your proposed solution is
> working well because the gate is actually supposed to be white. What you've
> created here is sort of a riff on very early two-colour film systems which
> used red and green; they sort of work in the same way this sort of works,
> giving a quasi-reasonable rendition of skin tones and foliage.
> If you have other situations, though, where you have subjects that are
> supposed to be just green, they'll end up cyan. I'm surprised it isn't more
> visible on the plants.
> I'd gently suggest that something like Resolve (free version available for
> major platforms) which allows you to get creative in how and where this
> stuff is applied.
>

Why you must always recommend alternative crappy solutions here?


> P
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Re: [FFmpeg-user] Seeking smart decomb filter

[Paul B Mahol]

On Thu, Jan 21, 2021 at 2:34 PM Mark Filipak (ffmpeg) 
wrote:

> On 01/21/2021 07:54 AM, Paul B Mahol wrote:
> > On Thu, Jan 21, 2021 at 1:51 PM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 01/21/2021 07:31 AM, Paul B Mahol wrote:
> >>> On Thu, Jan 21, 2021 at 12:51 PM Mark Filipak (ffmpeg) <
> >> markfili...@bog.us>
> >>> wrote:
> >>>
> >>>> On 01/20/2021 04:50 PM, Paul B Mahol wrote:
> >>>>> On Wed, Jan 20, 2021 at 10:45 PM Mark Filipak (ffmpeg) <
> >>>> markfili...@bog.us>
> >>>>> wrote:
> >>>>>> On 01/20/2021 06:17 AM, Paul B Mahol wrote:
> >>>>>>> Motion compensation does not work that way.
> >>>>>>
> >>>>>> Thank you, Paul. Yes, I knew that. I'm not seeking motion
> >> compensation.
> >>>>>> Kindly reread my filter
> >>>>>> features and suggest what comes closest. What I seek simply works on
> >>>>>> pixels. I'm relying on your
> >>>>>> experience (and perhaps the experiences of others here).
> >>>>>
> >>>>> Have you tried nnedi or estdif filters? Both are intra only
> >>>> deinterlacers.
> >>>>
> >>>> Thank you for the guidance. I'm trying estdif first because it seems
> >>>> simpler, but it doesn't do the
> >>>> good thing. It does decomb very effectively, but it does it by
> aligning
> >>>> (shifting) the edges in
> >>>> field'2' to the edges in field'1' (which are not shifted). Aligning
> with
> >>>> field'1' produces judder
> >>>> (or in my case, doesn't eliminate judder). What I want it to do is
> align
> >>>> both field'1' edges and
> >>>> field'2' edges to the median (i.e. half way between the input's
> edges).
> >> I
> >>>> do realize that such a
> >>>> method would introduce combing along the edges of the frame, but
> that's
> >>>> okay.
> >>>>
> >>>> Here's the command I used to transcode 24FPS to 60FPS:
> >>>> ffmpeg -i 24[1][2].mkv -map 0 -filter_complex "telecine=pattern=5,
> >>>> split[1][2],
> >>>> [1]select=not(eq(mod(n\,5)\,2))[3], [2]select=eq(mod(n\,5)\,2),
> >>>> estdif=mode=frame[4],
> >>>> [3][4]interleave" -codec:v libx265 -x265-params "crf=16:qcomp=0.60"
> >>>> -codec:a copy -codec:s copy
> >>>> 24[1][2]-to-60[1][1][estdif=mode=frame(1~2)][2][2].mkv
> >>>>
> >>>> I'll try nnedi. Perhaps I can figure it out.
> >>>>
> >>>
> >>> Do not even try.
> >>> Mentioned filters work only with interlaced frames, while your use case
> >> is
> >>> completely invalid and frowned upon.
> >>> I deeply regret replying to this thread.
> >>
> >> I did get nnedi to work but with similar results to estdif.
> >>
> >> Kindly educate me: What do you mean by "interlaced frames"? Aren't all
> >> frames that come out of the
> >> decoder and into the filter chain interlaced?
> >>
> >> Why do you say that my use case is invalid? telecine=pattern=5 works
> very
> >> well -- there's no judder
> >> at all! And if, for the combed frame, I use pp=linblenddeint, it's even
> >> better, far better than what
> >> either the NVIDIA GPU or the TV does when fed a 24FPS source and of
> course
> >> far, far better than
> >> telecine=pattern=23.
> >>
> >> Please educate me. Why are my stunning results invalid?
> >>
> >
> > Because you are trying to interpolate new frames using telecine filter.
>
> I am? I apologize for not being clear...
>
> The source is 24FPS. 'telecine=pattern=5' makes a 60FPS target with frames
> 0 & 1 == source frame 0,
> and frames 3 & 4 == source frame 1. Target frame 2 (i.e. the target frame
> in the middle) is combed
> of course. What I'm trying to do is decomb it. How is that interpolating
> new frames? Educate me. And
> if it is interpolating new frames, what's wrong with that? -- It looks
> stunning on a 60Hz TV whereas
> 24FPS fed to a 60Hz TV has serious judder.
>

You want to decomb telecined frames. But telecine already used existing
frames to create combed frames.
So you actually seek frame that is between these two fields in combed
frame, you cant creat

Re: [FFmpeg-user] Seeking smart decomb filter

[Paul B Mahol]

On Thu, Jan 21, 2021 at 1:51 PM Mark Filipak (ffmpeg) 
wrote:

> On 01/21/2021 07:31 AM, Paul B Mahol wrote:
> > On Thu, Jan 21, 2021 at 12:51 PM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> On 01/20/2021 04:50 PM, Paul B Mahol wrote:
> >>> On Wed, Jan 20, 2021 at 10:45 PM Mark Filipak (ffmpeg) <
> >> markfili...@bog.us>
> >>> wrote:
> >>>> On 01/20/2021 06:17 AM, Paul B Mahol wrote:
> >>>>> Motion compensation does not work that way.
> >>>>
> >>>> Thank you, Paul. Yes, I knew that. I'm not seeking motion
> compensation.
> >>>> Kindly reread my filter
> >>>> features and suggest what comes closest. What I seek simply works on
> >>>> pixels. I'm relying on your
> >>>> experience (and perhaps the experiences of others here).
> >>>
> >>> Have you tried nnedi or estdif filters? Both are intra only
> >> deinterlacers.
> >>
> >> Thank you for the guidance. I'm trying estdif first because it seems
> >> simpler, but it doesn't do the
> >> good thing. It does decomb very effectively, but it does it by aligning
> >> (shifting) the edges in
> >> field'2' to the edges in field'1' (which are not shifted). Aligning with
> >> field'1' produces judder
> >> (or in my case, doesn't eliminate judder). What I want it to do is align
> >> both field'1' edges and
> >> field'2' edges to the median (i.e. half way between the input's edges).
> I
> >> do realize that such a
> >> method would introduce combing along the edges of the frame, but that's
> >> okay.
> >>
> >> Here's the command I used to transcode 24FPS to 60FPS:
> >> ffmpeg -i 24[1][2].mkv -map 0 -filter_complex "telecine=pattern=5,
> >> split[1][2],
> >> [1]select=not(eq(mod(n\,5)\,2))[3], [2]select=eq(mod(n\,5)\,2),
> >> estdif=mode=frame[4],
> >> [3][4]interleave" -codec:v libx265 -x265-params "crf=16:qcomp=0.60"
> >> -codec:a copy -codec:s copy
> >> 24[1][2]-to-60[1][1][estdif=mode=frame(1~2)][2][2].mkv
> >>
> >> I'll try nnedi. Perhaps I can figure it out.
> >>
> >
> > Do not even try.
> > Mentioned filters work only with interlaced frames, while your use case
> is
> > completely invalid and frowned upon.
> > I deeply regret replying to this thread.
>
> I did get nnedi to work but with similar results to estdif.
>
> Kindly educate me: What do you mean by "interlaced frames"? Aren't all
> frames that come out of the
> decoder and into the filter chain interlaced?
>
> Why do you say that my use case is invalid? telecine=pattern=5 works very
> well -- there's no judder
> at all! And if, for the combed frame, I use pp=linblenddeint, it's even
> better, far better than what
> either the NVIDIA GPU or the TV does when fed a 24FPS source and of course
> far, far better than
> telecine=pattern=23.
>
> Please educate me. Why are my stunning results invalid?
>

Because you are trying to interpolate new frames using telecine filter.


>
> Regards,
> Mark.
>
> >> Thanks again,
> >> Mark.
> >>
> >>>> Regards,
> >>>> Mark.
> >>>>
> >>>>> On Wed, Jan 20, 2021 at 2:45 AM Mark Filipak (ffmpeg) <
> >>>> markfili...@bog.us>
> >>>>> wrote:
> >>>>>
> >>>>>> Hello All,
> >>>>>>
> >>>>>> I seek a decomb filter that operates on H/2 number of line pairs:
> >> lines
> >>>> i
> >>>>>> & i+1 (where i=0..H/2-1),
> >>>>>> by aligning edges in the two lines at x = delta-x(edge)/2 (i.e. the
> >>>>>> median). The ideal filter would
> >>>>>> differentiate overall motion due to panning versus local motion due
> to
> >>>>>> local object motion.
> >>>>>>
> >>>>>> Pan-combing on the left & right edges is acceptable but it would be
> a
> >>>>>> great bonus if the filter
> >>>>>> performed blend on those edges with the blending radius equal to
> >>>> one-half
> >>>>>> the pan-comb.
> >>>>>>
> >>>>>> Vertical, line-pair-to-line-pair processing is not needed or
> desired.
> >>>>>>
> >>>>>> Frame-to-frame processing is not needed or desired.
> >>>>>>
> >>>>>> I have been trying a great number of filters based on their names
> and
> >>>> what
> >>>>>> is in the documentation.
> >>>>>> The best I've found so far is pp=linblenddeint but it obviously
> >> doesn't
> >>>>>> align edges via the edge
> >>>>>> median and it obviously isn't aware of pan-combing versus
> >>>>>> object-motion-combing.
> >>>>>>
> >>>>>> Any suggestions are very welcome. If I find this filter I will post
> a
> >>>> link
> >>>>>> to a 60FPS telecined
> >>>>>> video that will amaze everyone -- better than anything that Cuda can
> >> do.
> >>>>>>
> >>>>>> -Mark.
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Re: [FFmpeg-user] Seeking smart decomb filter

[Paul B Mahol]

On Thu, Jan 21, 2021 at 1:31 PM Paul B Mahol  wrote:

>
>
> On Thu, Jan 21, 2021 at 12:51 PM Mark Filipak (ffmpeg) 
> wrote:
>
>> On 01/20/2021 04:50 PM, Paul B Mahol wrote:
>> > On Wed, Jan 20, 2021 at 10:45 PM Mark Filipak (ffmpeg) <
>> markfili...@bog.us>
>> > wrote:
>> >> On 01/20/2021 06:17 AM, Paul B Mahol wrote:
>> >>> Motion compensation does not work that way.
>> >>
>> >> Thank you, Paul. Yes, I knew that. I'm not seeking motion compensation.
>> >> Kindly reread my filter
>> >> features and suggest what comes closest. What I seek simply works on
>> >> pixels. I'm relying on your
>> >> experience (and perhaps the experiences of others here).
>> >
>> > Have you tried nnedi or estdif filters? Both are intra only
>> deinterlacers.
>>
>> Thank you for the guidance. I'm trying estdif first because it seems
>> simpler, but it doesn't do the
>> good thing. It does decomb very effectively, but it does it by aligning
>> (shifting) the edges in
>> field'2' to the edges in field'1' (which are not shifted). Aligning with
>> field'1' produces judder
>> (or in my case, doesn't eliminate judder). What I want it to do is align
>> both field'1' edges and
>> field'2' edges to the median (i.e. half way between the input's edges). I
>> do realize that such a
>> method would introduce combing along the edges of the frame, but that's
>> okay.
>>
>> Here's the command I used to transcode 24FPS to 60FPS:
>> ffmpeg -i 24[1][2].mkv -map 0 -filter_complex "telecine=pattern=5,
>> split[1][2],
>> [1]select=not(eq(mod(n\,5)\,2))[3], [2]select=eq(mod(n\,5)\,2),
>> estdif=mode=frame[4],
>> [3][4]interleave" -codec:v libx265 -x265-params "crf=16:qcomp=0.60"
>> -codec:a copy -codec:s copy
>> 24[1][2]-to-60[1][1][estdif=mode=frame(1~2)][2][2].mkv
>>
>> I'll try nnedi. Perhaps I can figure it out.
>>
>
> Do not even try.
> Mentioned filters work only with interlaced frames, while your use case is
> completely invalid and frowned upon.
> I deeply regret replying to this thread.
>

Also, the only way to "invent" new frames is to use either motion
interpolation via optical flow or via blocks interpolation like with
minterpolate filter.


>
>
>>
>> Thanks again,
>> Mark.
>>
>> >> Regards,
>> >> Mark.
>> >>
>> >>> On Wed, Jan 20, 2021 at 2:45 AM Mark Filipak (ffmpeg) <
>> >> markfili...@bog.us>
>> >>> wrote:
>> >>>
>> >>>> Hello All,
>> >>>>
>> >>>> I seek a decomb filter that operates on H/2 number of line pairs:
>> lines
>> >> i
>> >>>> & i+1 (where i=0..H/2-1),
>> >>>> by aligning edges in the two lines at x = delta-x(edge)/2 (i.e. the
>> >>>> median). The ideal filter would
>> >>>> differentiate overall motion due to panning versus local motion due
>> to
>> >>>> local object motion.
>> >>>>
>> >>>> Pan-combing on the left & right edges is acceptable but it would be a
>> >>>> great bonus if the filter
>> >>>> performed blend on those edges with the blending radius equal to
>> >> one-half
>> >>>> the pan-comb.
>> >>>>
>> >>>> Vertical, line-pair-to-line-pair processing is not needed or desired.
>> >>>>
>> >>>> Frame-to-frame processing is not needed or desired.
>> >>>>
>> >>>> I have been trying a great number of filters based on their names and
>> >> what
>> >>>> is in the documentation.
>> >>>> The best I've found so far is pp=linblenddeint but it obviously
>> doesn't
>> >>>> align edges via the edge
>> >>>> median and it obviously isn't aware of pan-combing versus
>> >>>> object-motion-combing.
>> >>>>
>> >>>> Any suggestions are very welcome. If I find this filter I will post a
>> >> link
>> >>>> to a 60FPS telecined
>> >>>> video that will amaze everyone -- better than anything that Cuda can
>> do.
>> >>>>
>> >>>> -Mark.
>>
>> ___
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>> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>
>
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Re: [FFmpeg-user] Seeking smart decomb filter

[Paul B Mahol]

On Thu, Jan 21, 2021 at 12:51 PM Mark Filipak (ffmpeg) 
wrote:

> On 01/20/2021 04:50 PM, Paul B Mahol wrote:
> > On Wed, Jan 20, 2021 at 10:45 PM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >> On 01/20/2021 06:17 AM, Paul B Mahol wrote:
> >>> Motion compensation does not work that way.
> >>
> >> Thank you, Paul. Yes, I knew that. I'm not seeking motion compensation.
> >> Kindly reread my filter
> >> features and suggest what comes closest. What I seek simply works on
> >> pixels. I'm relying on your
> >> experience (and perhaps the experiences of others here).
> >
> > Have you tried nnedi or estdif filters? Both are intra only
> deinterlacers.
>
> Thank you for the guidance. I'm trying estdif first because it seems
> simpler, but it doesn't do the
> good thing. It does decomb very effectively, but it does it by aligning
> (shifting) the edges in
> field'2' to the edges in field'1' (which are not shifted). Aligning with
> field'1' produces judder
> (or in my case, doesn't eliminate judder). What I want it to do is align
> both field'1' edges and
> field'2' edges to the median (i.e. half way between the input's edges). I
> do realize that such a
> method would introduce combing along the edges of the frame, but that's
> okay.
>
> Here's the command I used to transcode 24FPS to 60FPS:
> ffmpeg -i 24[1][2].mkv -map 0 -filter_complex "telecine=pattern=5,
> split[1][2],
> [1]select=not(eq(mod(n\,5)\,2))[3], [2]select=eq(mod(n\,5)\,2),
> estdif=mode=frame[4],
> [3][4]interleave" -codec:v libx265 -x265-params "crf=16:qcomp=0.60"
> -codec:a copy -codec:s copy
> 24[1][2]-to-60[1][1][estdif=mode=frame(1~2)][2][2].mkv
>
> I'll try nnedi. Perhaps I can figure it out.
>

Do not even try.
Mentioned filters work only with interlaced frames, while your use case is
completely invalid and frowned upon.
I deeply regret replying to this thread.


>
> Thanks again,
> Mark.
>
> >> Regards,
> >> Mark.
> >>
> >>> On Wed, Jan 20, 2021 at 2:45 AM Mark Filipak (ffmpeg) <
> >> markfili...@bog.us>
> >>> wrote:
> >>>
> >>>> Hello All,
> >>>>
> >>>> I seek a decomb filter that operates on H/2 number of line pairs:
> lines
> >> i
> >>>> & i+1 (where i=0..H/2-1),
> >>>> by aligning edges in the two lines at x = delta-x(edge)/2 (i.e. the
> >>>> median). The ideal filter would
> >>>> differentiate overall motion due to panning versus local motion due to
> >>>> local object motion.
> >>>>
> >>>> Pan-combing on the left & right edges is acceptable but it would be a
> >>>> great bonus if the filter
> >>>> performed blend on those edges with the blending radius equal to
> >> one-half
> >>>> the pan-comb.
> >>>>
> >>>> Vertical, line-pair-to-line-pair processing is not needed or desired.
> >>>>
> >>>> Frame-to-frame processing is not needed or desired.
> >>>>
> >>>> I have been trying a great number of filters based on their names and
> >> what
> >>>> is in the documentation.
> >>>> The best I've found so far is pp=linblenddeint but it obviously
> doesn't
> >>>> align edges via the edge
> >>>> median and it obviously isn't aware of pan-combing versus
> >>>> object-motion-combing.
> >>>>
> >>>> Any suggestions are very welcome. If I find this filter I will post a
> >> link
> >>>> to a 60FPS telecined
> >>>> video that will amaze everyone -- better than anything that Cuda can
> do.
> >>>>
> >>>> -Mark.
>
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Re: [FFmpeg-user] Seeking smart decomb filter

[Paul B Mahol]

On Wed, Jan 20, 2021 at 10:45 PM Mark Filipak (ffmpeg) 
wrote:

> On 01/20/2021 06:17 AM, Paul B Mahol wrote:
> > Motion compensation does not work that way.
>
> Thank you, Paul. Yes, I knew that. I'm not seeking motion compensation.
> Kindly reread my filter
> features and suggest what comes closest. What I seek simply works on
> pixels. I'm relying on your
> experience (and perhaps the experiences of others here).
>

Have you tried nnedi or estdif filters? Both are intra only deinterlacers.


>
> Regards,
> Mark.
>
> > On Wed, Jan 20, 2021 at 2:45 AM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> Hello All,
> >>
> >> I seek a decomb filter that operates on H/2 number of line pairs: lines
> i
> >> & i+1 (where i=0..H/2-1),
> >> by aligning edges in the two lines at x = delta-x(edge)/2 (i.e. the
> >> median). The ideal filter would
> >> differentiate overall motion due to panning versus local motion due to
> >> local object motion.
> >>
> >> Pan-combing on the left & right edges is acceptable but it would be a
> >> great bonus if the filter
> >> performed blend on those edges with the blending radius equal to
> one-half
> >> the pan-comb.
> >>
> >> Vertical, line-pair-to-line-pair processing is not needed or desired.
> >>
> >> Frame-to-frame processing is not needed or desired.
> >>
> >> I have been trying a great number of filters based on their names and
> what
> >> is in the documentation.
> >> The best I've found so far is pp=linblenddeint but it obviously doesn't
> >> align edges via the edge
> >> median and it obviously isn't aware of pan-combing versus
> >> object-motion-combing.
> >>
> >> Any suggestions are very welcome. If I find this filter I will post a
> link
> >> to a 60FPS telecined
> >> video that will amaze everyone -- better than anything that Cuda can do.
> >>
> >> -Mark.
>
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Re: [FFmpeg-user] Seeking smart decomb filter

[Paul B Mahol]

Motion compensation does not work that way.

On Wed, Jan 20, 2021 at 2:45 AM Mark Filipak (ffmpeg) 
wrote:

> Hello All,
>
> I seek a decomb filter that operates on H/2 number of line pairs: lines i
> & i+1 (where i=0..H/2-1),
> by aligning edges in the two lines at x = delta-x(edge)/2 (i.e. the
> median). The ideal filter would
> differentiate overall motion due to panning versus local motion due to
> local object motion.
>
> Pan-combing on the left & right edges is acceptable but it would be a
> great bonus if the filter
> performed blend on those edges with the blending radius equal to one-half
> the pan-comb.
>
> Vertical, line-pair-to-line-pair processing is not needed or desired.
>
> Frame-to-frame processing is not needed or desired.
>
> I have been trying a great number of filters based on their names and what
> is in the documentation.
> The best I've found so far is pp=linblenddeint but it obviously doesn't
> align edges via the edge
> median and it obviously isn't aware of pan-combing versus
> object-motion-combing.
>
> Any suggestions are very welcome. If I find this filter I will post a link
> to a 60FPS telecined
> video that will amaze everyone -- better than anything that Cuda can do.
>
> -Mark.
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Re: [FFmpeg-user] HW Acceleration 101? 2-Up Streaming from RTSP-->ffmpeg-->YouTube

[Paul B Mahol]

Try using more optimized filters like, hstack/vstack. nullsrc and two
overlays when there is no alpha is pointless.

On Mon, Jan 18, 2021 at 7:49 PM Steven Kan  wrote:

> I’ve been using fffmpeg as a “relay station” for a few years to pull RTSP
> streams from several IP cameras and push them to YouTube, such as my 24/7
> BeeCam:
>
> https://www.youtube.com/channel/UCE0jx2Z6qbc5Co8x8Kyisag/live <
> https://www.youtube.com/channel/UCE0jx2Z6qbc5Co8x8Kyisag/live>
>
> Command is of the form:
>
> ./ffmpeg -re -thread_queue_size 512 -rtsp_transport tcp -i "rtsp://
> anonymous:password@192.168.1.50:554  anonymous:password@192.168.1.50:554>" -vcodec copy -acodec copy -t
> 01:00:00 -f flv "rtmp://a.rtmp.youtube.com/live2/
> ”
>
> Because I’m deliberately just relaying the packets and _not_ doing any
> transcoding, the CPU utilization is remarkably low, and independent of
> camera resolution. I can run 3 instances of ffmpeg on Raspbian/Raspberry Pi
> 3B+ and each uses only about 10% of the CPU, despite each pushing a 5 MP
> camera stream.
>
> This works very well; no problems here!
>
> But now I want to do a “2-Up” live stream of two different cameras,
> side-by-side. Here’s an archive from last night (waiting for a mating pair
> of Barn Owls to move in):
>
> https://www.youtube.com/watch?v=GDN2MjPwn0Q=youtu.be <
> https://www.youtube.com/watch?v=GDN2MjPwn0Q=youtu.be>
>
> The cameras are each outputting 1920 x 1080 @ 25 fps.
>
> Now that I’m actually encoding, I need a lot more CPU/GPU. I’m running
> this in Win10 Pro/64 on an HP Microserver with an AMD Opteron X3418
> Quad-Core, and the CPU runs at about ~65-80% while the integrated GPU runs
> at about ~55%.
>
> C:\Program Files\ffmpeg\bin> .\ffmpeg.exe -re -thread_queue_size 1024 -i
> rtsp://anonymous:password@192.168.1.47:554  anonymous:password@192.168.1.47:554> -i rtsp://
> anonymous:password@192.168.1.50:554  anonymous:password@192.168.1.50:554> -vcodec h264_amf -acodec copy -t
> 01:47:02 -filter_complex "nullsrc=size=3840x1080 [base]; [0:v]
> setpts=PTS-STARTPTS, scale=1920x1080 [upperleft]; [1:v]
> setpts=PTS-STARTPTS, scale=1920x1080 [upperright]; [base][upperleft]
> overlay=shortest=1 [tmp1]; [tmp1][upperright] overlay=shortest=1:x=1920" -f
> flv "rtmp://a.rtmp.youtube.com/live2/my-youtube-streaming-key  a.rtmp.youtube.com/live2/my-youtube-streaming-key>”
>
> ffmpeg version 2020-12-09-git-e5119a-essentials_build-www.gyan.dev <
> http://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-llvm --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. 62.100 / 56. 62.100
>   libavcodec 58.115.102 / 58.115.102
>   libavformat58. 65.100 / 58. 65.100
>   libavdevice58. 11.103 / 58. 11.103
>   libavfilter 7. 92.100 /  7. 92.100
>   libswscale  5.  8.100 /  5.  8.100
>   libswresample   3.  8.100 /  3.  8.100
>   libpostproc55.  8.100 / 55.  8.100
> Input #0, rtsp, from 'rtsp://anonymous:password@192.168.1.47:554':
> 
>   Metadata:
> title   : Media Server
>   Duration: N/A, start: 0.08, bitrate: N/A
> Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709, progressive),
> 1920x1080, 25 fps, 25 tbr, 90k tbn, 180k tbc
> Input #1, rtsp, from 'rtsp://anonymous:password@192.168.1.50:554':
> 
>   Metadata:
> title   : Media Server
>   Duration: N/A, start: 0.10, bitrate: N/A
> Stream #1:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, 100
> tbr, 90k tbn, 180k tbc
> Stream #1:1: Audio: aac (LC), 8000 Hz, mono, fltp
> Stream mapping:
>   Stream #0:0 (h264) -> setpts
>   Stream #1:0 (h264) -> setpts
>   overlay -> Stream #0:0 (h264_amf)
>   Stream #1:1 -> #0:1 (copy)
> Press [q] to stop, [?] for help
> [rtsp @ 0199e58e14c0] Thread message queue blocking; consider raising
> the thread_queue_size option (current value: 8)
> [rtsp @ 0199e586d9c0] max delay reached. need to consume packet
> [rtsp @ 0199e586d9c0] RTP: missed 157 packets
> [swscaler @ 0199e5cc5b80] deprecated pixel format used, make sure you
> did set range correctly
> 

Re: [FFmpeg-user] tinterlace broken - SAR & DAR wrong

[Paul B Mahol]

On Mon, Jan 4, 2021 at 9:21 PM Mark Filipak (ffmpeg) 
wrote:

> On 01/04/2021 02:45 PM, pdr0 wrote:
> > Mark Filipak (ffmpeg) wrote
> >>
> >> You wrote: "What's wrong with using setsar filter after tinterlace?"
> >>
> >> I tried that from the git-go. I just reran it.
> >>
> >> ffmpeg -report -i "source 720x480 [SAR 8x9 DAR 4x3] 29.97 fps.VOB"
> >> -filter_complex "separatefields,
> >> shuffleframes=0 1 2 4 3 6 5 7 8 9, split[Ain][Bin],
> >>
> [Ain]select=not(eq(mod(n\,10)\,2)+eq(mod(n\,10)\,3)+eq(mod(n\,10)\,6)+eq(mod(n\,10)\,7))[Aout],
> >>
> [Bin]select=eq(mod(n\,10)\,2)+eq(mod(n\,10)\,3)+eq(mod(n\,10)\,6)+eq(mod(n\,10)\,7),
> >> tinterlace,
> >> setsar=sar=8/9, bwdif=mode=send_field:deint=all, decimate=cycle=2[Bout],
> >> [Aout][Bout]interleave,
> >> tinterlace, setsar=sar=8/9, bwdif=mode=send_frame:deint=interlaced"
> >> -codec:a copy -codec:s copy -dn
> >> output.mkv
> >>
> >> Putting setsar after tinterlace doesn't work. I did investigate that
> but I
> >> can't recall why -- I
> >> think it's because DAR is then wrong. The report is below.
> >
> >
> > You can't interleave images with different dimensions
> >
> > Aout has separated fields, to 720x240 , but Bout is 720x480
>
> [Bout] is 720x240: I'm using 'mode=send_field' in 'bwdif=mode=send_field',
> and the following
> 'decimate' doesn't change that. The problem is that 'tinterleave' is
> marking the output as "SAR =
> 16:9" instead of "8:9" (which is what it should be and should never be
> changed).
>
> SAR cannot change unless DAR/PAR changes (which it doesn't). Look:
>
> SAR = DAR/PAR = (8:3)/(720:240) = (8:3)/(6:2) = 8:9
> SAR = DAR/PAR = (4:3)/(720:480) = (4:3)/(3:2) = 8:9
>
> The current tinterlace filter is setting SAR = 16:9. That is just plain
> wrong.
>

That is just invalid statement.


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Re: [FFmpeg-user] tinterlace broken - SAR & DAR wrong

[Paul B Mahol]

On Sun, Jan 3, 2021 at 9:28 PM Paul B Mahol  wrote:

>
>
> On Sun, Jan 3, 2021 at 9:25 PM Mark Filipak (ffmpeg) 
> wrote:
>
>> On 01/03/2021 10:24 AM, Paul B Mahol wrote:
>> > On Sun, Jan 3, 2021 at 11:45 AM Mark Filipak (ffmpeg) <
>> markfili...@bog.us>
>> > wrote:
>> >
>> >> Version N-100546-g2c6f532e0a (latest git build)
>> >>
>> >> Summary of the bug:
>> >> tinterlace works but sets SAR & DAR to bogus values.
>> >>
>> >> The package is here:
>> >> https://www.dropbox.com/t/0WABzvetTF1eT7E4
>> >>
>> >> Everything needed to reproduce and source video and my result and log
>> are
>> >> in the package.
>> >>
>> >>
>> > The change of sar is intended behavior.
>>
>> Okay.
>> Given this:
>> ffmpeg -report -i source.mkv -vf "tinterlace=mode=merge" -c:a copy -c:s
>> copy -dn tinterlace.mkv
>> where source.mpv is this:
>> "Input #0, matroska,webm, from 'source.mkv':
>> "  Metadata:
>> "ENCODER : Lavf58.65.100
>> "  Duration: 00:00:01.00, start: 0.00, bitrate: 857 kb/s
>> "Stream #0:0, 4, 1/1000: Video: h264 (High), yuv420p(tv, smpte170m,
>> progressive), 720x240 [SAR
>> 8:9 DAR 8:3], 59.94 fps, 59.94 tbr, 1k tbn, 119.88 tbc (default)"
>>
>> tinterlace.mkv should be this: 720x480 [SAR 8:9 DAR 4:3]
>> but instead is this:
>> "Output #0, matroska, to 'tinterlace.mkv':
>> "  Metadata:
>> "encoder : Lavf58.65.100
>> "Stream #0:0, 0, 1/1000: Video: h264 (H264 / 0x34363248), yuv420p(tv,
>> smpte170m, top coded first
>> (swapped)), 720x480 [SAR 16:9 DAR 8:3], q=2-31, 29.97 fps, 1k tbn
>> (default)"
>>
>> The discrepancy appears to be causing problems with succeeding filters:
>> "[Parsed_interleave_7 @ 0170340dc000] Parameters for input link
>> input1 (size 720x480, SAR 16:9)
>> do not match the corresponding output link parameters (720x240, SAR 8:9)
>> "[Parsed_interleave_7 @ 0170340dc000] Failed to configure output pad
>> on Parsed_interleave_7
>> "Error reinitializing filters!
>> "Failed to inject frame into filter network: Invalid argument
>> "Error while processing the decoded data for stream #0:0
>> "Conversion failed!"
>>
>> The succeeding filters are shuffleframes, split, select, tinterlace,
>> bwdif, decimate, interleave,
>> tinterlace, and bwdif. The problem appears to be provoked by interleave.
>>
>
> What's wrong with using setsar filter after tinterlace?
>

Also you can use weave filter as alternative to tinterlace=merge


>
>>
>> I promise to follow this message with the full command line and full
>> report if you promise not to
>> ask me why I'm doing such seemingly crazy processing.  :-)
>>
>> Honestly, I'm not trying to be coy. I'm simply trying to keep this
>> use-case simple and prevent this
>> thread from getting off topic. The problem appears to be with
>> tinterlace's SAR.
>>
>> Regards,
>> Mark.
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>> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>
>
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Re: [FFmpeg-user] tinterlace broken - SAR & DAR wrong

[Paul B Mahol]

On Sun, Jan 3, 2021 at 9:25 PM Mark Filipak (ffmpeg) 
wrote:

> On 01/03/2021 10:24 AM, Paul B Mahol wrote:
> > On Sun, Jan 3, 2021 at 11:45 AM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> Version N-100546-g2c6f532e0a (latest git build)
> >>
> >> Summary of the bug:
> >> tinterlace works but sets SAR & DAR to bogus values.
> >>
> >> The package is here:
> >> https://www.dropbox.com/t/0WABzvetTF1eT7E4
> >>
> >> Everything needed to reproduce and source video and my result and log
> are
> >> in the package.
> >>
> >>
> > The change of sar is intended behavior.
>
> Okay.
> Given this:
> ffmpeg -report -i source.mkv -vf "tinterlace=mode=merge" -c:a copy -c:s
> copy -dn tinterlace.mkv
> where source.mpv is this:
> "Input #0, matroska,webm, from 'source.mkv':
> "  Metadata:
> "ENCODER : Lavf58.65.100
> "  Duration: 00:00:01.00, start: 0.00, bitrate: 857 kb/s
> "Stream #0:0, 4, 1/1000: Video: h264 (High), yuv420p(tv, smpte170m,
> progressive), 720x240 [SAR
> 8:9 DAR 8:3], 59.94 fps, 59.94 tbr, 1k tbn, 119.88 tbc (default)"
>
> tinterlace.mkv should be this: 720x480 [SAR 8:9 DAR 4:3]
> but instead is this:
> "Output #0, matroska, to 'tinterlace.mkv':
> "  Metadata:
> "encoder : Lavf58.65.100
> "Stream #0:0, 0, 1/1000: Video: h264 (H264 / 0x34363248), yuv420p(tv,
> smpte170m, top coded first
> (swapped)), 720x480 [SAR 16:9 DAR 8:3], q=2-31, 29.97 fps, 1k tbn
> (default)"
>
> The discrepancy appears to be causing problems with succeeding filters:
> "[Parsed_interleave_7 @ 0170340dc000] Parameters for input link input1
> (size 720x480, SAR 16:9)
> do not match the corresponding output link parameters (720x240, SAR 8:9)
> "[Parsed_interleave_7 @ 0170340dc000] Failed to configure output pad
> on Parsed_interleave_7
> "Error reinitializing filters!
> "Failed to inject frame into filter network: Invalid argument
> "Error while processing the decoded data for stream #0:0
> "Conversion failed!"
>
> The succeeding filters are shuffleframes, split, select, tinterlace,
> bwdif, decimate, interleave,
> tinterlace, and bwdif. The problem appears to be provoked by interleave.
>

What's wrong with using setsar filter after tinterlace?


>
> I promise to follow this message with the full command line and full
> report if you promise not to
> ask me why I'm doing such seemingly crazy processing.  :-)
>
> Honestly, I'm not trying to be coy. I'm simply trying to keep this
> use-case simple and prevent this
> thread from getting off topic. The problem appears to be with tinterlace's
> SAR.
>
> Regards,
> Mark.
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Re: [FFmpeg-user] tinterlace broken - SAR & DAR wrong

[Paul B Mahol]

On Sun, Jan 3, 2021 at 11:45 AM Mark Filipak (ffmpeg) 
wrote:

> Version N-100546-g2c6f532e0a (latest git build)
>
> Summary of the bug:
> tinterlace works but sets SAR & DAR to bogus values.
>
> The package is here:
> https://www.dropbox.com/t/0WABzvetTF1eT7E4
>
> Everything needed to reproduce and source video and my result and log are
> in the package.
>
>
The change of sar is intended behavior.



> Thank you,
> Mark.
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Re: [FFmpeg-user] xmedian questions

[Paul B Mahol]

On Fri, Jan 1, 2021 at 4:08 AM Mark Filipak (ffmpeg) 
wrote:

> On 12/31/2020 09:50 PM, list+ffmpeg-u...@jdlh.com wrote:
> > On 2020-12-31 17:25, Mark Filipak (ffmpeg) wrote:
> >> 1, Median? Really? (Or maybe really average?)
> >>
> >> For inputs = 239+15+15+15:
> >> median = (239+15)/2 = 127
> >> average = (239+15+15+15)/4 = 71
> >> For inputs = 239+239+239+15:
> >> median = (239+15)/2 = 127 (again)
> >> average = (239+239+239+15)/4 = 183
> >
> > It jumped out at me that the expressions you are computing for "median"
> don't match the definition I
> > am used to. It looks like you are computing the average of the largest
> and smallest samples. I
> > understand "median" to be a value such that half the samples are greater
> than or equal to, and half
> > are less than equal to, the value.
> >
> > For inputs = 239+15+15+15:
> > median = 15 (239, 15 >= 15; 15, 15 <= 15)
> > For inputs = 239+239+239+15:
> > median = 239 (239, 239 >= 239; 239, 15 <= 15)
> >
> > So, how to you define "median"? And how does FFmpeg?
> >
> >   —Jim DeLaHunt
>
> Thank you, Jim. You are correct. I reckon I was thinking of some sort of
> mean value -- there are
> several versions of mean.
>
> Median: The middle number in a given sequence of numbers, taken as the
> average of the two middle
> numbers when the sequence has an even number of numbers.
>
> Hmmm... What sort of weighting is that?
>
> For inputs = 239+15+15+15:
> median = (15+15)/2 = 15
> average = (239+15+15+15)/4 = 71
> For inputs = 239+239+239+15:
> median = (239+239)/2 = 239
> average = (239+239+239+15)/4 = 183
>
> As you see, the median winds up being either 15 or 239.
> So, what's the function of 'percentile'?
>

Input pixels are sorted first.
percentile says where in sorted list to pick pixel value.


> And why is "inputs=2" not allowed?
>

Because it is same as mix=2 filter

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Re: [FFmpeg-user] Decimate question

[Paul B Mahol]

On Wed, Dec 30, 2020 at 6:44 PM Mark Filipak (ffmpeg) 
wrote:

> On 12/30/2020 03:35 AM, Paul B Mahol wrote:
> > On Wed, Dec 30, 2020 at 6:08 AM Mark Filipak (ffmpeg) <
> markfili...@bog.us>
> > wrote:
> >
> >> Decimate will drop a repeated frame in each sequence of 5 (=default)
> >> frames. I assume the dropped
> >> frame can be any 1 of the 5. Is there a way to discover which of the 5
> >> frames it dropped?
> >>
> >
> > -v debug
>
> May I make a feature request? Could it be added as ... is it called side
> data?
>
> I haven't used side data yet so I'm not quite sure what it is and how it's
> used.
>
> The frame # would need to be readable by follow-on logic inside
> expressions.
>
> This is for processing videos that have previously been telecined, a frame
> dropped, and a frame
> repeated, in various sequences. The sequence of the video destruction is
> not important. What's
> important is the phase of the decimation within the modulo-5 in the video
> that remains. In 4 of the
> 5 cases (phases), it's actually possible to almost completely recover the
> original video (prior to
> the wrecking). 4 out of 5 cases involve the loss of just 1 field in 10.
> The 5th case involves the
> total loss of a whole frame and is not recoverable (but since that case
> would have an amazing amount
> of judder, it's unlikely to be found in the wild). However, I have found
> other cases of this video
> species in the wild.
>

Sorry, but you make nonsense entries, very often and as expected.



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