Re: [FFmpeg-user] ffmpeg - compressing an mp4 creates a video distortion (horizontal stretch)

[William C Bonner]

On Thu, Jun 13, 2024 at 1:06 PM  wrote:

> I have a mp4 file that was created by concatenating several videos taken
> with a cellphone. Some of the videos were taken horizontally and others
> vertically.
>
> The concatenated video plays fine but when I try to compress it so it can
> be
> placed in a web site I have the following problem with the segments that
> were taken vertically:
>
> In the original file (before compression) the segments taken vertically
> look
> good (no distortion) but they have black areas on the sides, which I
> believe
> is expected as the width of the vertical video is smaller than the width of
> the horizontal ones.
>
> In the compressed file, the segments coming from the vertical videos are
> distorted. The compression removed the black areas on each side and
> stretched the frame to have the full width of final video. In other words,
> it made the width of the vertical segments equal to the width of the
> horizontal ones, creating the distortion (stretching).
>
> Why is this happening and how can I avoid it?
>
> I uploaded the video to youtube to check if the compression they use
> results
> in the same problem, and it did. So it looks like the ffmpeg command I am
> using is similar to what YouTube does.
>
> Here is the command I used:
>
> Ffmpeg  -i concatenated_Final.mp4 -c:v libx264 -preset slow -crf 20 -r 30
> -c:a copy -pix_fmt yuv420p concatenated_Final_Compressed.mp4
>
> I tried with crf 20 and crf 31 but the problem was the same.
>
>
 I believe that the concatenation commands don't automatically deal with
differing video types, Different framerate, resolution,etc will all cause
unknown results. I don't see the ffmpeg command of how you created the
concatenated video.

When you are playing a vertical video and getting black bars on the sides,
that's the player recognizing the shape of the video and adding the bars.
Similar if you get bars at the top and bottom if your video is a different
perspective than the screen.

You need to preprocess each file to be the same format, then you can
concatenate them together.
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[FFmpeg-user] ffmpeg - compressing an mp4 creates a video distortion (horizontal stretch)

[iburman]

I have a mp4 file that was created by concatenating several videos taken
with a cellphone. Some of the videos were taken horizontally and others
vertically.

The concatenated video plays fine but when I try to compress it so it can be
placed in a web site I have the following problem with the segments that
were taken vertically:

 

In the original file (before compression) the segments taken vertically look
good (no distortion) but they have black areas on the sides, which I believe
is expected as the width of the vertical video is smaller than the width of
the horizontal ones. 

 

In the compressed file, the segments coming from the vertical videos are
distorted. The compression removed the black areas on each side and
stretched the frame to have the full width of final video. In other words,
it made the width of the vertical segments equal to the width of the
horizontal ones, creating the distortion (stretching).

 

Why is this happening and how can I avoid it?

 

I uploaded the video to youtube to check if the compression they use results
in the same problem, and it did. So it looks like the ffmpeg command I am
using is similar to what YouTube does.

 

Here is the command I used:

Ffmpeg  -i concatenated_Final.mp4 -c:v libx264 -preset slow -crf 20 -r 30
-c:a copy -pix_fmt yuv420p concatenated_Final_Compressed.mp4

 

I tried with crf 20 and crf 31 but the problem was the same.

 

Thanks

 

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Re: [FFmpeg-user] FFmpeg-devel signup and patch status

[Thilo Borgmann via ffmpeg-user]

Hi,


I tried to sign up on FFmpeg-devel, but the reset password is not reaching
me.


your mail is not subscribed to ffmpeg-devel (yet).
Did you check your spam for registration mails?
GMail behaves very strict sometimes, trying another mail address might be worth 
it in the long run if this keeps being a problem for you.



The patch I would submit is just this patch but based on version 7.0 and
with an updated doc/protocols.texi. I think that was the reason this was
not merged before?

https://patchwork.ffmpeg.org/project/ffmpeg/patch/20190925185708.70924-1-unique.will.mar...@gmail.com/


-Thilo

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[FFmpeg-user] FFmpeg-devel signup and patch status

[Renato Santos]

Hello,

I tried to sign up on FFmpeg-devel, but the reset password is not reaching
me.

The patch I would submit is just this patch but based on version 7.0 and
with an updated doc/protocols.texi. I think that was the reason this was
not merged before?

https://patchwork.ffmpeg.org/project/ffmpeg/patch/20190925185708.70924-1-unique.will.mar...@gmail.com/

If anyone could help, I would appreciate it.
Thank you,
-- 
YAGNI,
Renato CRON
@renato_cron 
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[FFmpeg-user] ffmpeg and ffprobe 7.0 cannot read video files from the Sony A7SIII camera

[Jean Vézina]

Good evening,

I have upgraded  ffmpeg (and ffprobe) from 6.1.1 to 7.0 and the Sony A7SIII 
video files can no longer be read.
I can supply a test file if required.

Regards,

Jean


Example with 6.1.1 (correct)

ffprobe C0033.MP4
ffprobe version 6.1.1 Copyright (c) 2007-2023 the FFmpeg developers
  built with Apple clang version 15.0.0 (clang-1500.3.9.4)
  configuration: --prefix='/opt/homebrew/Cellar/ffmpeg@6/6.1.1_1' 
--enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= 
--host-ldflags='-Wl,-ld_classic' --enable-ffplay --enable-gnutls --enable-gpl 
--enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d 
--enable-libharfbuzz --enable-libjxl --enable-libmp3lame --enable-libopus 
--enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy 
--enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract 
--enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis 
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 
--enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig 
--enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopenjpeg --enable-libopenvino 
--enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg 
--disable-libjack --disable-indev=jack --enable-v
 ideotoolbox --enable-audiotoolbox --enable-neon
  libavutil  58. 29.100 / 58. 29.100
  libavcodec 60. 31.102 / 60. 31.102
  libavformat60. 16.100 / 60. 16.100
  libavdevice60.  3.100 / 60.  3.100
  libavfilter 9. 12.100 /  9. 12.100
  libswscale  7.  5.100 /  7.  5.100
  libswresample   4. 12.100 /  4. 12.100
  libpostproc57.  3.100 / 57.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x124f06260] st: 0 edit list: 1 Missing key frame 
while searching for timestamp: 2002
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x124f06260] st: 0 edit list 1 Cannot find an index 
entry before timestamp: 2002.
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C0033.MP4':
  Metadata:
major_brand : XAVC
minor_version   : 17506303
compatible_brands: XAVCmp42nrasiso6
creation_time   : 2024-05-25T17:54:46.00Z
  Duration: 00:00:35.04, start: 0.00, bitrate: 92011 kb/s
  Stream #0:0[0x1](und): Video: hevc (Main 10) (hvc1 / 0x31637668), 
yuv420p10le(tv, bt2020nc/bt2020/arib-std-b67), 3840x2160 [SAR 1:1 DAR 16:9], 
75423 kb/s, 59.94 fps, 59.94 tbr, 60k tbn (default)
Metadata:
  creation_time   : 2024-05-25T17:54:46.00Z
  handler_name: Video Media Handler
  vendor_id   : [0][0][0][0]
  encoder : HEVC Coding
  Stream #0:1[0x2](und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, 2 
channels, s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2024-05-25T17:54:46.00Z
  handler_name: Sound Media Handler
  vendor_id   : [0][0][0][0]
  Stream #0:2[0x3](und): Data: none (rtmd / 0x646D7472), 9329 kb/s (default)
Metadata:
  creation_time   : 2024-05-25T17:54:46.00Z
  handler_name: Timed Metadata Media Handler
  timecode: 00:13:53:18
Unsupported codec with id 0 for input stream 2

Example with version 7.0 (failed)

 ffprobe C0033.MP4
ffprobe version 7.0 Copyright (c) 2007-2024 the FFmpeg developers
  built with Apple clang version 15.0.0 (clang-1500.3.9.4)
  configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/7.0_1 --enable-shared 
--enable-pthreads --enable-version3 --cc=clang --host-cflags= 
--host-ldflags='-Wl,-ld_classic' --enable-ffplay --enable-gnutls --enable-gpl 
--enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d 
--enable-libharfbuzz --enable-libjxl --enable-libmp3lame --enable-libopus 
--enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy 
--enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract 
--enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis 
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 
--enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig 
--enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopenjpeg --enable-libopenvino 
--enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg 
--disable-libjack --disable-indev=jack --enable-videoto
 olbox --enable-audiotoolbox --enable-neon
  libavutil  59.  8.100 / 59.  8.100
  libavcodec 61.  3.100 / 61.  3.100
  libavformat61.  1.100 / 61.  1.100
  libavdevice61.  1.100 / 61.  1.100
  libavfilter10.  1.100 / 10.  1.100
  libswscale  8.  1.100 /  8.  1.100
  libswresample   5.  1.100 /  5.  1.100
  libpostproc58.  1.100 / 58.  1.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x11cf04880] st: 0 edit list: 1 Missing key frame 
while searching for timestamp: 2002
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x11cf04880] st: 0 edit list 1 Cannot find an index 
entry before timestamp: 2002.
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x11cf04880] infe: 

[FFmpeg-user] ffmpeg built without liblensfun

[Chris Wilkinson via ffmpeg-user]

Hi all, I've been trying to figure out how to correct lens distortion in video 
for a very long time, but tonight have discovered that ffmpeg has a plugin for 
the excellent lensfun library but the ubuntu PPA packaged versions for 
22.04 are built with --disable-liblensfun (why oh why?).
Does anyone know if there is a ffmpeg PPA for Ubuntu 22.04 or a binary version 
as a snap etc that is built including liblensfun?
Best regards,Chris W, NZ
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 24.05.24 um 01:23 schrieb John Hawkinson:

Zathras: It appears your assumption (6) that the files are "exactly the
same" because the spectrograms are the same is not correct.

I downloaded https://www2.cs.uic.edu/~i101/SoundFiles/BabyElephantWalk60.wav
as a test file and converted it with the freeoconvert.com online converter,
and ffprobe reports the following:

% ffprobe BabyElephantWalk60.mov
...
   Stream #0:1[0x2]: Audio: aac (LC) (mp4a / 0x6134706D), 22050 Hz, mono,
fltp, 72 kb/s (default)
 Metadata:
   handler_name: SoundHandler
   vendor_id   : [0][0][0][0]

That is, the online converter you specify has converted the audio to AAC.
That is, you should be able to use simply


so you tell me the moron pretends all the time his downlaod has a wav 
stream and it hasn't? jesus



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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Moritz Barsnick via ffmpeg-user]

On Fri, May 24, 2024 at 02:23:16 +0200, Ferdi Scholten wrote:
> > what is this supposed to be?
> > 16 Bit with 44,1 kHz (CD) are sufficient.
> > So, we have PCM_S16BE and PCM_S16LE.
> >
> > Whats the exact command to get it done?
>
> Youtube simply does not accept lossless audio in a video (at all). You have
> to encode according to their specifications. They explicitly say to use AAC
> in .mov containers, so do not use anything else as it will be not be
> accepted.

As far as I understand these help pages, this is a recommendation.
Indeed, YouTube also uses ffmpeg (so they say) for processing their
files, and accepts quite a lot. They may reject some formats and codecs
by filter - e.g. to avoid uncompressed input - but it's a matter of
trial and error and experience.

(Streaming to YouTube "live" is even more tricky.)

> You can only upload lossless audio (as .wav or .flac) as extra language for
> an existing uploaded video

Is that so?

BTW, the pages saying that PCM in WAV is accepted may either be
enormously outdated, or referring to YouTube Music. You really need to
be careful with trusting search results.

> this lossless audio will be converted to a lossy
> format by Youtube that will fit the videostream (usually AAC or Opus).

THIS is the important point. While providing lossless input in order to
not lose quality even before submission is nice idea (if it works), you
will never get lossless formats back when playing the media from
YouTube.

And lossless video is not really bandwidth-friendly anyway, but that
wasn't the question obviously.

Cheers,
Moritz
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Paul B Mahol]

On Fri, May 24, 2024 at 1:12 AM Zathras Yes 
wrote:

> Hi,
>
> as i have mentioned to Mr. Reindl, the online-converter
> https://www.freeconvert.com/wav-to-mov  does the trick.
>
> ffmpeg should be able to do the same, I suppose.
> May be, I am in error, may be not...
>
> 1. make a wav-files (16 Bit, 44.1 kHz) "sound.wav"
> 2. use the mentioned online converter from "sound.wav" to "upload.mov"
> 3. Upload this "upload.mov" to youtube.
> 4. Go to the youtube-link and download the file "download.mov"
> 5. extract the sound file out of "download.mov" --> "extracted.wav"
> 6. Compare the original file "sound.wav" with "extracted.wav" with its
> spectrogramms. (Audacity or Spectral Layers for instance)
> --> et voila the files are exactly the same!
>

"Audacity or Spectral Layers" spectrograms does not mean much, at least
half of data, the phase, is not shown in typical spectrograms.

Two .wav files are exactly the same if cmp UNIX tool returns empty output
when given paths to two .wav files as argmunt to cmp tool.

The AAC SLS (The lossless variant of AAC) is completely different sub-codec
for AAC, and very marginally supported.



> So, it does work somehow.
> For this, I do not understand the length of the discussion (especially with
> Mr. Reindl), that it does not work, because it is not supposed to work?
> But it does -  so I wonder why it is such a big deal here.
>
>
> On Fri, May 24, 2024 at 1:02 AM Andrew Randrianasulu <
> randrianas...@gmail.com> wrote:
>
> > пт, 24 мая 2024 г., 01:37 Reindl Harald :
> >
> > >
> > >
> > > Am 24.05.24 um 00:34 schrieb Zathras Yes:
> > > > Yes, I know that. Just add a picture and it's fine.
> > >
> > > MOV with PCM/WAV is never fine
> > >
> >
> > from compatibility with older software? yes. Youtube probably does not
> want
> > to serve relatively huge lossless uncompressed audio files, even if that
> > was perfectly possible.
> >
> > but for experimenting you can use ffbrobe on wav file and try to put -c:a
> > pcm_s32le for example if ffprobe indicated this format for your wav.
> >
> >
> >
> >
> > > > Please don't repeat all this over and over, because it is tiresome.
> > >
> > > if you won't be an idiot you would understand it by telling once
> > >
> > > > For this now, I will not further answer to you!
> > >
> > > i hope so
> > >
> > > > On Fri, May 24, 2024 at 12:32 AM Reindl Harald <
> h.rei...@thelounge.net
> > >
> > > > wrote:
> > > >
> > > >>
> > > >>
> > > >> Am 24.05.24 um 00:29 schrieb Zathras Yes:
> > > >>> Supported file formats:
> > > >>>
> > > >>>  -
> > > >>>  - PCM audio in WAV container
> > > >>>
> > > >>>
> > > >>> So, it seems possible to put a wav into a youtube-video, right?
> > > >>>
> > > >>> How to do it?
> > > >>
> > > >> https://support.google.com/youtube/troubleshooter/2888402?hl=en
> > > >>
> > > >> Note: Audio files, like MP3, WAV, or PCM files, can’t be uploaded to
> > > >> create a YouTube video. You can use video editing software to
> convert
> > > >> your audio file to a video. Audio files can only be uploaded as
> extra
> > > >> languages for your video.
> > >
> > > ___
> > > 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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> >
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Ferdi Scholten]

Hi,

what is this supposed to be?
16 Bit with 44,1 kHz (CD) are sufficient.
So, we have PCM_S16BE and PCM_S16LE.

Whats the exact command to get it done?


Thx



Youtube simply does not accept lossless audio in a video (at all). You 
have to encode according to their specifications. They explicitly say to 
use AAC in .mov containers, so do not use anything else as it will be 
not be accepted.


You can only upload lossless audio (as .wav or .flac) as extra language 
for an existing uploaded video, this lossless audio will be converted to 
a lossy format by Youtube that will fit the videostream (usually AAC or 
Opus).

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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[John Hawkinson]

Zathras: It appears your assumption (6) that the files are "exactly the
same" because the spectrograms are the same is not correct.

I downloaded https://www2.cs.uic.edu/~i101/SoundFiles/BabyElephantWalk60.wav
as a test file and converted it with the freeoconvert.com online converter,
and ffprobe reports the following:

% ffprobe BabyElephantWalk60.mov
...
  Stream #0:1[0x2]: Audio: aac (LC) (mp4a / 0x6134706D), 22050 Hz, mono,
fltp, 72 kb/s (default)
Metadata:
  handler_name: SoundHandler
  vendor_id   : [0][0][0][0]

That is, the online converter you specify has converted the audio to AAC.
That is, you should be able to use simply

ffmpeg -i picture.jpg -i music.wav upload.mov

and accept the results.

Alternatively, discussion in this thread suggests you can use the FLAC
encoding, which is lossless.

I agree that the responses you received to your inquiry were rude and
unprofessional and have no place on this mailing list and I am sorry you
had to deal with them, and I do not understand who administers this list or
how to resolve those issues.
--
jh...@alum.mit.edu
John Hawkinson


On Thu, May 23, 2024 at 7:12 PM Zathras Yes 
wrote:

> Hi,
>
> as i have mentioned to Mr. Reindl, the online-converter
> https://www.freeconvert.com/wav-to-mov  does the trick.
>
> ffmpeg should be able to do the same, I suppose.
> May be, I am in error, may be not...
>
> 1. make a wav-files (16 Bit, 44.1 kHz) "sound.wav"
> 2. use the mentioned online converter from "sound.wav" to "upload.mov"
> 3. Upload this "upload.mov" to youtube.
> 4. Go to the youtube-link and download the file "download.mov"
> 5. extract the sound file out of "download.mov" --> "extracted.wav"
> 6. Compare the original file "sound.wav" with "extracted.wav" with its
> spectrogramms. (Audacity or Spectral Layers for instance)
> --> et voila the files are exactly the same!
>
> So, it does work somehow.
> For this, I do not understand the length of the discussion (especially with
> Mr. Reindl), that it does not work, because it is not supposed to work?
> But it does -  so I wonder why it is such a big deal here.
>
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Hi,

as i have mentioned to Mr. Reindl, the online-converter
https://www.freeconvert.com/wav-to-mov  does the trick.

ffmpeg should be able to do the same, I suppose.
May be, I am in error, may be not...

1. make a wav-files (16 Bit, 44.1 kHz) "sound.wav"
2. use the mentioned online converter from "sound.wav" to "upload.mov"
3. Upload this "upload.mov" to youtube.
4. Go to the youtube-link and download the file "download.mov"
5. extract the sound file out of "download.mov" --> "extracted.wav"
6. Compare the original file "sound.wav" with "extracted.wav" with its
spectrogramms. (Audacity or Spectral Layers for instance)
--> et voila the files are exactly the same!

So, it does work somehow.
For this, I do not understand the length of the discussion (especially with
Mr. Reindl), that it does not work, because it is not supposed to work?
But it does -  so I wonder why it is such a big deal here.


On Fri, May 24, 2024 at 1:02 AM Andrew Randrianasulu <
randrianas...@gmail.com> wrote:

> пт, 24 мая 2024 г., 01:37 Reindl Harald :
>
> >
> >
> > Am 24.05.24 um 00:34 schrieb Zathras Yes:
> > > Yes, I know that. Just add a picture and it's fine.
> >
> > MOV with PCM/WAV is never fine
> >
>
> from compatibility with older software? yes. Youtube probably does not want
> to serve relatively huge lossless uncompressed audio files, even if that
> was perfectly possible.
>
> but for experimenting you can use ffbrobe on wav file and try to put -c:a
> pcm_s32le for example if ffprobe indicated this format for your wav.
>
>
>
>
> > > Please don't repeat all this over and over, because it is tiresome.
> >
> > if you won't be an idiot you would understand it by telling once
> >
> > > For this now, I will not further answer to you!
> >
> > i hope so
> >
> > > On Fri, May 24, 2024 at 12:32 AM Reindl Harald  >
> > > wrote:
> > >
> > >>
> > >>
> > >> Am 24.05.24 um 00:29 schrieb Zathras Yes:
> > >>> Supported file formats:
> > >>>
> > >>>  -
> > >>>  - PCM audio in WAV container
> > >>>
> > >>>
> > >>> So, it seems possible to put a wav into a youtube-video, right?
> > >>>
> > >>> How to do it?
> > >>
> > >> https://support.google.com/youtube/troubleshooter/2888402?hl=en
> > >>
> > >> Note: Audio files, like MP3, WAV, or PCM files, can’t be uploaded to
> > >> create a YouTube video. You can use video editing software to convert
> > >> your audio file to a video. Audio files can only be uploaded as extra
> > >> languages for your video.
> >
> > ___
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> > 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] ffmpeg wav lossless to youtube mov?

[Andrew Randrianasulu]

пт, 24 мая 2024 г., 01:37 Reindl Harald :

>
>
> Am 24.05.24 um 00:34 schrieb Zathras Yes:
> > Yes, I know that. Just add a picture and it's fine.
>
> MOV with PCM/WAV is never fine
>

from compatibility with older software? yes. Youtube probably does not want
to serve relatively huge lossless uncompressed audio files, even if that
was perfectly possible.

but for experimenting you can use ffbrobe on wav file and try to put -c:a
pcm_s32le for example if ffprobe indicated this format for your wav.




> > Please don't repeat all this over and over, because it is tiresome.
>
> if you won't be an idiot you would understand it by telling once
>
> > For this now, I will not further answer to you!
>
> i hope so
>
> > On Fri, May 24, 2024 at 12:32 AM Reindl Harald 
> > wrote:
> >
> >>
> >>
> >> Am 24.05.24 um 00:29 schrieb Zathras Yes:
> >>> Supported file formats:
> >>>
> >>>  -
> >>>  - PCM audio in WAV container
> >>>
> >>>
> >>> So, it seems possible to put a wav into a youtube-video, right?
> >>>
> >>> How to do it?
> >>
> >> https://support.google.com/youtube/troubleshooter/2888402?hl=en
> >>
> >> Note: Audio files, like MP3, WAV, or PCM files, can’t be uploaded to
> >> create a YouTube video. You can use video editing software to convert
> >> your audio file to a video. Audio files can only be uploaded as extra
> >> languages for your video.
>
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 24.05.24 um 00:34 schrieb Zathras Yes:

Yes, I know that. Just add a picture and it's fine.


MOV with PCM/WAV is never fine


Please don't repeat all this over and over, because it is tiresome.


if you won't be an idiot you would understand it by telling once


For this now, I will not further answer to you!


i hope so


On Fri, May 24, 2024 at 12:32 AM Reindl Harald 
wrote:




Am 24.05.24 um 00:29 schrieb Zathras Yes:

Supported file formats:

 -
 - PCM audio in WAV container


So, it seems possible to put a wav into a youtube-video, right?

How to do it?


https://support.google.com/youtube/troubleshooter/2888402?hl=en

Note: Audio files, like MP3, WAV, or PCM files, can’t be uploaded to
create a YouTube video. You can use video editing software to convert
your audio file to a video. Audio files can only be uploaded as extra
languages for your video.


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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 24.05.24 um 00:31 schrieb Zathras Yes:

Well, I am not ignorant, just stating the facts, especially about the
online-converter.

What you are writing contradicts with the facts here:
It's not a matter of liking or not, it's a fact.

*
Now, I tell you what:

1. make a wav-files (16 Bit, 44.1 kHz) "sound.wav"
2. use the mentioned online converter from "sound.wav" to "upload.mov"
3. Upload this "upload.mov" to youtube.
4. Go to the youtube-link and download the file "download.mov"
5. extract the sound file out of "download.mov" --> "extracted.wav"
6. Compare the original file "sound.wav" with "extracted.wav" with its
spectrogramms. (Audacity or Spectral Layers for instance)
--> et voila the files are exactly the same!

Now what?


now it's bullshit and not a valid MOV file

* AAC audio in MOV container
* Note: Audio files, like MP3, WAV, or PCM files,
  can’t be uploaded to create a YouTube video.
  You can use video editing software to convert
  your audio file to a video. Audio files can only
  be uploaded as extra languages for your video.

a f**ng WAV audio-track donÄrt belong into a MOV-CONTAINER
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Yes, I know that. Just add a picture and it's fine.

Please don't repeat all this over and over, because it is tiresome.

For this now, I will not further answer to you!

Thx.

On Fri, May 24, 2024 at 12:32 AM Reindl Harald 
wrote:

>
>
> Am 24.05.24 um 00:29 schrieb Zathras Yes:
> > Supported file formats:
> >
> > -
> > - PCM audio in WAV container
> >
> >
> > So, it seems possible to put a wav into a youtube-video, right?
> >
> > How to do it?
>
> https://support.google.com/youtube/troubleshooter/2888402?hl=en
>
> Note: Audio files, like MP3, WAV, or PCM files, can’t be uploaded to
> create a YouTube video. You can use video editing software to convert
> your audio file to a video. Audio files can only be uploaded as extra
> languages for your video.
>
>
> ___
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> ffmpeg-user@ffmpeg.org
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 24.05.24 um 00:29 schrieb Zathras Yes:

Supported file formats:

-
- PCM audio in WAV container


So, it seems possible to put a wav into a youtube-video, right?

How to do it?


https://support.google.com/youtube/troubleshooter/2888402?hl=en

Note: Audio files, like MP3, WAV, or PCM files, can’t be uploaded to 
create a YouTube video. You can use video editing software to convert 
your audio file to a video. Audio files can only be uploaded as extra 
languages for your video.



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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Well, I am not ignorant, just stating the facts, especially about the
online-converter.

What you are writing contradicts with the facts here:
It's not a matter of liking or not, it's a fact.

*
Now, I tell you what:

1. make a wav-files (16 Bit, 44.1 kHz) "sound.wav"
2. use the mentioned online converter from "sound.wav" to "upload.mov"
3. Upload this "upload.mov" to youtube.
4. Go to the youtube-link and download the file "download.mov"
5. extract the sound file out of "download.mov" --> "extracted.wav"
6. Compare the original file "sound.wav" with "extracted.wav" with its
spectrogramms. (Audacity or Spectral Layers for instance)
--> et voila the files are exactly the same!

Now what?

On Fri, May 24, 2024 at 12:29 AM Reindl Harald 
wrote:

>
> Am 24.05.24 um 00:25 schrieb Zathras Yes:
> > Well, why being so rude?
>
> because you are ignorant
>
> > Than would you be so kind to explain, why the online converter does the
> > trick?
>
> it don't
>
>
> > I would expect, that ffmpeg could do the same, right?
> >
>
> a MOV-CONTAINER can't contain WAV/PCM
> youtube says: AAC audio in MOV container
>
>
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Supported file formats:

   -
   - PCM audio in WAV container


So, it seems possible to put a wav into a youtube-video, right?

How to do it?

On Fri, May 24, 2024 at 12:26 AM Reindl Harald 
wrote:

>
>
> Am 24.05.24 um 00:22 schrieb Zathras Yes:
> > Now, I tell you what:
> >
> > 1. make a wav-files (16 Bit, 44.1 kHz) "sound.wav"
>
> i did
>
> > 2. use the mentioned online converter from "sound.wav" to "upload.mov"
>
> i did and i posted you the error screen
>
> > 3. Upload this "upload.mov" to youtube
>
> impossible because it won't output anything
>
> > Now what?
>
> now understand that a MOV-CONTAINER must not contain WAV/PCM audio
> and since this isn't wikipedia educate yourself what a container is
>
> https://support.google.com/youtube/answer/4603579?hl=en
>
> GOD DAMNED: AAC audio in MOV container
>
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 24.05.24 um 00:25 schrieb Zathras Yes:

Well, why being so rude?


because you are ignorant


Than would you be so kind to explain, why the online converter does the
trick?


it don't



I would expect, that ffmpeg could do the same, right?



a MOV-CONTAINER can't contain WAV/PCM
youtube says: AAC audio in MOV container


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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 24.05.24 um 00:22 schrieb Zathras Yes:

Now, I tell you what:

1. make a wav-files (16 Bit, 44.1 kHz) "sound.wav"


i did


2. use the mentioned online converter from "sound.wav" to "upload.mov"


i did and i posted you the error screen


3. Upload this "upload.mov" to youtube


impossible because it won't output anything


Now what?


now understand that a MOV-CONTAINER must not contain WAV/PCM audio
and since this isn't wikipedia educate yourself what a container is

https://support.google.com/youtube/answer/4603579?hl=en

GOD DAMNED: AAC audio in MOV container

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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Well, why being so rude?

Than would you be so kind to explain, why the online converter does the
trick?
I would expect, that ffmpeg could do the same, right?



On Fri, May 24, 2024 at 12:22 AM Reindl Harald 
wrote:

>
>
> Am 24.05.24 um 00:17 schrieb Reindl Harald:
> >
> >
> > Am 24.05.24 um 00:12 schrieb Zathras Yes:
> >> Ok, instead of repeating here over and over:
> >>
> >> First question: Is video with lossless sound possible for youtube? Yes
> or
> >> No answer please.
> >
> > no - and whatever you upload to youtube will be finally re-encoded
> > anyways - you won't find any WAV audio on the whole youtube world
>
> https://support.google.com/youtube/answer/4603579?hl=en
>
> Audio file guidelines
>
> The following guidelines are for audio tracks that you provide to
> YouTube. These guidelines describe the formatting specifications that
> yield the highest quality for playing audio on YouTube and for matching
> your audio tracks to the audio tracks of user-uploaded videos. Note that
> an audio track would only be played back on YouTube if you have opted to
> include that track in YouTube's AudioSwap program. Generally, we
> recommend that you upload the highest quality audio possible.
>
>  Supported file formats:
>  MP3 audio in MP3/WAV container
>  PCM audio in WAV container
>  AAC audio in MOV container
>  FLAC audio
>
> --
>
> "wav lossless to youtube mov" is bullshit - period
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Now, I tell you what:

1. make a wav-files (16 Bit, 44.1 kHz) "sound.wav"
2. use the mentioned online converter from "sound.wav" to "upload.mov"
3. Upload this "upload.mov" to youtube.
4. Go to the youtube-link and download the file "download.mov"
5. extract the sound file out of "download.mov" --> "extracted.wav"
6. Compare the original file "sound.wav" with "extracted.wav" with its
spectrogramms. (Audacity or Spectral Layers for instance)
--> et voila the files are exactly the same!

Now what?


On Fri, May 24, 2024 at 12:18 AM Reindl Harald 
wrote:

>
>
> Am 24.05.24 um 00:12 schrieb Zathras Yes:
> > Ok, instead of repeating here over and over:
> >
> > First question: Is video with lossless sound possible for youtube? Yes or
> > No answer please.
>
> no - and whatever you upload to youtube will be finally re-encoded
> anyways - you won't find any WAV audio on the whole youtube world
> ___
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 24.05.24 um 00:17 schrieb Reindl Harald:



Am 24.05.24 um 00:12 schrieb Zathras Yes:

Ok, instead of repeating here over and over:

First question: Is video with lossless sound possible for youtube? Yes or
No answer please.


no - and whatever you upload to youtube will be finally re-encoded 
anyways - you won't find any WAV audio on the whole youtube world


https://support.google.com/youtube/answer/4603579?hl=en

Audio file guidelines

The following guidelines are for audio tracks that you provide to 
YouTube. These guidelines describe the formatting specifications that 
yield the highest quality for playing audio on YouTube and for matching 
your audio tracks to the audio tracks of user-uploaded videos. Note that 
an audio track would only be played back on YouTube if you have opted to 
include that track in YouTube's AudioSwap program. Generally, we 
recommend that you upload the highest quality audio possible.


Supported file formats:
MP3 audio in MP3/WAV container
PCM audio in WAV container
AAC audio in MOV container
FLAC audio

--

"wav lossless to youtube mov" is bullshit - period
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Hi,

what is this supposed to be?
16 Bit with 44,1 kHz (CD) are sufficient.
So, we have PCM_S16BE and PCM_S16LE.

Whats the exact command to get it done?


Thx

On Fri, May 24, 2024 at 12:14 AM Andrew Randrianasulu <
randrianas...@gmail.com> wrote:

> пт, 24 мая 2024 г., 01:00 Reindl Harald :
>
> >
> >
> > Am 23.05.24 um 23:44 schrieb Zathras Yes:
> > > And yes, a conversion of the soundfiles is not wanted, exactly.
> > >
> > > In a picky manner of speaking, it is a conversion of one file to
> another,
> > > but let''s not go into this nitpicking here.
> >
> > again: "-acodec copy" is NOT a conversion - you are pumping a
> > unsupported audio-stream into a container with the result of an invalid
> > file
> >
> > this is not a ffmpeg problem - it's PEBCAK
> >
>
>
> /* ISO/IEC 23003-5 integer formats */
> { AV_CODEC_ID_PCM_S16BE,   MOV_MP4_IPCM_TAG  },
> { AV_CODEC_ID_PCM_S16LE,   MOV_MP4_IPCM_TAG  },
> { AV_CODEC_ID_PCM_S24BE,   MOV_MP4_IPCM_TAG  },
> { AV_CODEC_ID_PCM_S24LE,   MOV_MP4_IPCM_TAG  },
> { AV_CODEC_ID_PCM_S32BE,   MOV_MP4_IPCM_TAG  },
> { AV_CODEC_ID_PCM_S32LE,   MOV_MP4_IPCM_TAG  },
> /* ISO/IEC 23003-5 floating-point formats */
> { AV_CODEC_ID_PCM_F32BE,   MOV_MP4_FPCM_TAG  },
> { AV_CODEC_ID_PCM_F32LE,   MOV_MP4_FPCM_TAG  },
> { AV_CODEC_ID_PCM_F64BE,   MOV_MP4_FPCM_TAG  },
> { AV_CODEC_ID_PCM_F64LE,   MOV_MP4_FPCM_TAG  },
>
> from libavformat/movenc.c
>
> so I guess it should be muxable, but may be yt  not really likes increased
> total bitrate in this case, and whole thing (vid + aud) served now as  many
> little pieces mostly ...? I think ALAC also exist as experimental (strict
> -2) codec in ffmpeg.
>
>
> >
> > ___
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 24.05.24 um 00:12 schrieb Zathras Yes:

Ok, instead of repeating here over and over:

First question: Is video with lossless sound possible for youtube? Yes or
No answer please.


no - and whatever you upload to youtube will be finally re-encoded 
anyways - you won't find any WAV audio on the whole youtube world

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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Andrew Randrianasulu]

пт, 24 мая 2024 г., 01:00 Reindl Harald :

>
>
> Am 23.05.24 um 23:44 schrieb Zathras Yes:
> > And yes, a conversion of the soundfiles is not wanted, exactly.
> >
> > In a picky manner of speaking, it is a conversion of one file to another,
> > but let''s not go into this nitpicking here.
>
> again: "-acodec copy" is NOT a conversion - you are pumping a
> unsupported audio-stream into a container with the result of an invalid
> file
>
> this is not a ffmpeg problem - it's PEBCAK
>


/* ISO/IEC 23003-5 integer formats */
{ AV_CODEC_ID_PCM_S16BE,   MOV_MP4_IPCM_TAG  },
{ AV_CODEC_ID_PCM_S16LE,   MOV_MP4_IPCM_TAG  },
{ AV_CODEC_ID_PCM_S24BE,   MOV_MP4_IPCM_TAG  },
{ AV_CODEC_ID_PCM_S24LE,   MOV_MP4_IPCM_TAG  },
{ AV_CODEC_ID_PCM_S32BE,   MOV_MP4_IPCM_TAG  },
{ AV_CODEC_ID_PCM_S32LE,   MOV_MP4_IPCM_TAG  },
/* ISO/IEC 23003-5 floating-point formats */
{ AV_CODEC_ID_PCM_F32BE,   MOV_MP4_FPCM_TAG  },
{ AV_CODEC_ID_PCM_F32LE,   MOV_MP4_FPCM_TAG  },
{ AV_CODEC_ID_PCM_F64BE,   MOV_MP4_FPCM_TAG  },
{ AV_CODEC_ID_PCM_F64LE,   MOV_MP4_FPCM_TAG  },

from libavformat/movenc.c

so I guess it should be muxable, but may be yt  not really likes increased
total bitrate in this case, and whole thing (vid + aud) served now as  many
little pieces mostly ...? I think ALAC also exist as experimental (strict
-2) codec in ffmpeg.


>
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Ok, instead of repeating here over and over:

First question: Is video with lossless sound possible for youtube? Yes or
No answer please.

Second question: How to create such a files with ffmpeg? (The mentioned
online converter can do this) What's the ffmpeg command for it?


Thx



On Fri, May 24, 2024 at 12:00 AM Reindl Harald 
wrote:

>
>
> Am 23.05.24 um 23:44 schrieb Zathras Yes:
> > And yes, a conversion of the soundfiles is not wanted, exactly.
> >
> > In a picky manner of speaking, it is a conversion of one file to another,
> > but let''s not go into this nitpicking here.
>
> again: "-acodec copy" is NOT a conversion - you are pumping a
> unsupported audio-stream into a container with the result of an invalid
> file
>
> this is not a ffmpeg problem - it's PEBCAK
>
>
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 23.05.24 um 23:44 schrieb Zathras Yes:

And yes, a conversion of the soundfiles is not wanted, exactly.

In a picky manner of speaking, it is a conversion of one file to another,
but let''s not go into this nitpicking here.


again: "-acodec copy" is NOT a conversion - you are pumping a 
unsupported audio-stream into a container with the result of an invalid file


this is not a ffmpeg problem - it's PEBCAK


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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 23.05.24 um 23:44 schrieb Zathras Yes:

Sorry, but what you are saying here is simply plain *wrong* !

There *is* lossless music in mov.

The online converter does exactly that
https://www.freeconvert.com/wav-to-mov

You can try this for yourself, if you don't believe it.


i did and it IS NOT SUPPORTED

ffmpeg letg you combine whatever container, audio-format and 
video-format but taht don't make the result a valid file - hence youtube 
don't accept it - mov is a CONTAINER and it is NOT SUPPOSED to contain WAV

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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 23.05.24 um 23:30 schrieb Zathras Yes:

Hello,

I like to upload lossless sound to youtube.

ok, i forgot to add a picture.
so, we take this

ffmpeg.exe -i picture.jpg -i music.wav -acodec copy upload.mov

This gives a video with lossless audio, right?
(It's playing perfectly on my computer)
But youtube does not except his.

If the wav-File is put into this online-converter, it's working
https://www.freeconvert.com/wav-to-mov
This, youtube accepts


because the online-converter is smarter than you


Why would I use ffmpeg then?
Because the files are larger than 1 GB and I also prefer to do the
conversion myself.


you are *not* doing *any* conversion with "-acodec copy"
you are doing a dumb copy of a unsupported audio track

you can't have lossless audio in mov - period
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Hello,

I like to upload lossless sound to youtube.

ok, i forgot to add a picture.
so, we take this

ffmpeg.exe -i picture.jpg -i music.wav -acodec copy upload.mov

This gives a video with lossless audio, right?
(It's playing perfectly on my computer)
But youtube does not except his.

If the wav-File is put into this online-converter, it's working
https://www.freeconvert.com/wav-to-mov
This, youtube accepts

Why would I use ffmpeg then?
Because the files are larger than 1 GB and I also prefer to do the
conversion myself.

So how to do it with ffmpeg?
What is the command for it?

Thx.




On Thu, May 23, 2024 at 11:04 PM Reindl Harald 
wrote:

>
>
> Am 23.05.24 um 23:01 schrieb Reindl Harald:
> >
> >
> > Am 23.05.24 um 22:10 schrieb Zathras Yes:
> >> Hello,
> >>
> >> what is the command to put a wav-Sound file lossless into a mov-File in
> >> order to upload it correctly to youtube?
> >>
> >> For example, this does *not* work for youtube
> >>
> >> ffmpeg.exe -i music.wav -acodec copy upload.mov
> >
> > what makes you think you can use any random format?
> > wav has no place in mov/mpeg containers
>
> that above is pure nonsense anyways
> you could also do "mv music.wav upload.mov"
>
> why?
> because you only have a audio-track, tell ffmpeg to not touch it and
> write it into a file with a different extension - a little bit overhead
> using ffmpeg for a simple file-rename
>
> you need a video too for youtube
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 23.05.24 um 23:01 schrieb Reindl Harald:



Am 23.05.24 um 22:10 schrieb Zathras Yes:

Hello,

what is the command to put a wav-Sound file lossless into a mov-File in
order to upload it correctly to youtube?

For example, this does *not* work for youtube

ffmpeg.exe -i music.wav -acodec copy upload.mov


what makes you think you can use any random format?
wav has no place in mov/mpeg containers


that above is pure nonsense anyways
you could also do "mv music.wav upload.mov"

why?
because you only have a audio-track, tell ffmpeg to not touch it and 
write it into a file with a different extension - a little bit overhead 
using ffmpeg for a simple file-rename


you need a video too for youtube
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Re: [FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Reindl Harald]

Am 23.05.24 um 22:10 schrieb Zathras Yes:

Hello,

what is the command to put a wav-Sound file lossless into a mov-File in
order to upload it correctly to youtube?

For example, this does *not* work for youtube

ffmpeg.exe -i music.wav -acodec copy upload.mov


what makes you think you can use any random format?
wav has no place in mov/mpeg containers
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[FFmpeg-user] ffmpeg wav lossless to youtube mov?

[Zathras Yes]

Hello,

what is the command to put a wav-Sound file lossless into a mov-File in
order to upload it correctly to youtube?

For example, this does *not* work for youtube

ffmpeg.exe -i music.wav -acodec copy upload.mov

Thx
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Re: [FFmpeg-user] ffmpeg 6.1.1 fails to compile with mingw-w64 gcc 14

[Reindl Harald]

Am 17.05.24 um 19:20 schrieb Andrew Randrianasulu:

maybe - but it's nonsense to start a topic with an older version - the
first thing you have to do in case of compiler errors is to try the
latest version

why?

because it maybe already fixed and you waste everyobdys time playing
yround and asking questions about outdated software



But regressions exists, so I usually try with versions I have at hand, if
it fails - with newer and sometimes older versions too ...


you have https://ffmpeg.org/releases/ and GIT when it comes ot build 
from source *always* at hand - period
if you want to report something useful than note which versions you 
tried - but the first try to solve a problem is *always* try the latest 
version


on this mailing list you usually only have to report latest GIT which i 
find idiotic at my own - but in find the same for e release tarball from 
the last year

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Re: [FFmpeg-user] ffmpeg 6.1.1 fails to compile with mingw-w64 gcc 14

[Andrew Randrianasulu]

пт, 17 мая 2024 г., 20:02 Reindl Harald :

>
>
> Am 17.05.24 um 17:58 schrieb Biswapriyo Nath:
> >> what about trying the latest version instead 6.1.1 from last year when
> >> you use a compiler even newer than on Fedora 39?
> >
> > The same issue was reported with 7.0 release
> > https://trac.ffmpeg.org/ticket/11004. I am using msys2/mingw-w64
> > toolchain to compile ffmpeg. I guess the same compiler error should
> > happen with any other gcc 14 irrespective of software distribution.
>
> maybe - but it's nonsense to start a topic with an older version - the
> first thing you have to do in case of compiler errors is to try the
> latest version
>
> why?
>
> because it maybe already fixed and you waste everyobdys time playing
> yround and asking questions about outdated software
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пт, 17 мая 2024 г., 20:02 Reindl Harald :

>
>
> Am 17.05.24 um 17:58 schrieb Biswapriyo Nath:
> >> what about trying the latest version instead 6.1.1 from last year when
> >> you use a compiler even newer than on Fedora 39?
> >
> > The same issue was reported with 7.0 release
> > https://trac.ffmpeg.org/ticket/11004. I am using msys2/mingw-w64
> > toolchain to compile ffmpeg. I guess the same compiler error should
> > happen with any other gcc 14 irrespective of software distribution.
>
> maybe - but it's nonsense to start a topic with an older version - the
> first thing you have to do in case of compiler errors is to try the
> latest version
>
> why?
>
> because it maybe already fixed and you waste everyobdys time playing
> yround and asking questions about outdated software
>


But regressions exists, so I usually try with versions I have at hand, if
it fails - with newer and sometimes older versions too ...




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Re: [FFmpeg-user] ffmpeg 6.1.1 fails to compile with mingw-w64 gcc 14

[Reindl Harald]

Am 17.05.24 um 17:58 schrieb Biswapriyo Nath:

what about trying the latest version instead 6.1.1 from last year when
you use a compiler even newer than on Fedora 39?


The same issue was reported with 7.0 release
https://trac.ffmpeg.org/ticket/11004. I am using msys2/mingw-w64
toolchain to compile ffmpeg. I guess the same compiler error should
happen with any other gcc 14 irrespective of software distribution.


maybe - but it's nonsense to start a topic with an older version - the 
first thing you have to do in case of compiler errors is to try the 
latest version


why?

because it maybe already fixed and you waste everyobdys time playing 
yround and asking questions about outdated software

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Re: [FFmpeg-user] ffmpeg 6.1.1 fails to compile with mingw-w64 gcc 14

[Biswapriyo Nath]

> what about trying the latest version instead 6.1.1 from last year when
> you use a compiler even newer than on Fedora 39?

The same issue was reported with 7.0 release
https://trac.ffmpeg.org/ticket/11004. I am using msys2/mingw-w64
toolchain to compile ffmpeg. I guess the same compiler error should
happen with any other gcc 14 irrespective of software distribution.
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Re: [FFmpeg-user] ffmpeg 6.1.1 fails to compile with mingw-w64 gcc 14

[Reindl Harald]

Am 17.05.24 um 15:32 schrieb Biswapriyo Nath:

* gcc version: 14.1.0
* build options:
https://github.com/msys2/MINGW-packages/blob/master/mingw-w64-ffmpeg/PKGBUILD
* compiler error:

ffmpeg-6.1.1/libavutil/hwcontext_qsv.c:756:68: error: passing argument
3 of 'devmgr->lpVtbl->LockDevice' from incompatible pointer type
[-Wincompatible-pointer-types]
   756 | hr = IDirect3DDeviceManager9_LockDevice(devmgr,
device_handle, , TRUE);


what about trying the latest version instead 6.1.1 from last year when 
you use a compiler even newer than on Fedora 39?

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[FFmpeg-user] ffmpeg 6.1.1 fails to compile with mingw-w64 gcc 14

[Biswapriyo Nath]

* gcc version: 14.1.0
* build options:
https://github.com/msys2/MINGW-packages/blob/master/mingw-w64-ffmpeg/PKGBUILD
* compiler error:

ffmpeg-6.1.1/libavutil/hwcontext_qsv.c:756:68: error: passing argument
3 of 'devmgr->lpVtbl->LockDevice' from incompatible pointer type
[-Wincompatible-pointer-types]
  756 | hr = IDirect3DDeviceManager9_LockDevice(devmgr,
device_handle, , TRUE);
  |
^~~
  ||
  |
IDirect3DDevice9Ex **
ffmpeg-6.1.1/libavutil/hwcontext_qsv.c:756:68: note: expected
'IDirect3DDevice9 **' but argument is of type 'IDirect3DDevice9Ex **'

ffmpeg-6.1.1/libavutil/hwcontext_qsv.c:770:49: error: passing argument
2 of 'device->lpVtbl->GetDirect3D' from incompatible pointer type
[-Wincompatible-pointer-types]
  770 | hr = IDirect3DDevice9Ex_GetDirect3D(device, );
  | ^~~
  | |
  | IDirect3D9Ex **
ffmpeg-6.1.1/libavutil/hwcontext_qsv.c:770:49: note: expected
'IDirect3D9 **' but argument is of type 'IDirect3D9Ex **'

* downstream issue: https://github.com/msys2/MINGW-packages/issues/20891
* probable solution: https://github.com/msys2/MINGW-packages/pull/20930

Please feel free ask any further information I need to provide for
fixing the issue.

Thank you.
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Re: [FFmpeg-user] FFmpeg

[Reindl Harald]

Am 17.05.24 um 06:42 schrieb Keith Reno:

I have ffmpeg-7.0-full_build downloaded on my computer.  When I select ffmpeg 
the screen flashes but nothing loads.  Do I have the correct version or am I 
doing something wrong?


ffmpeg is a CLI program
https://en.wikipedia.org/wiki/Command-line_interface

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Re: [FFmpeg-user] FFmpeg

[Jim DeLaHunt]

Keith:

On 2024-05-16 21:42, Keith Reno wrote:

I have ffmpeg-7.0-full_build downloaded on my computer.  When I select ffmpeg 
the screen flashes but nothing loads.  Do I have the correct version or am I 
doing something wrong?

Keith


Welcome to the fun, fun world of FFmpeg!

Apologies for asking an elementary question, but, the FFmpeg program 
runs from the Command window. Are you opening a Command window and 
typing "ffmpeg" at the command prompt?


If these terms are unfamiliar, then a good next step might be to get a 
tutorial on using the Windows command prompt. Be sure to look for 
details on how to use quotes around command parameters. This list is not 
a good place to get that tutorial; people here mostly assume that 
everyone already knows how to use command-line interface programs, like 
FFmpeg and many others.


Best regards,
 —Jim DeLaHunt


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Re: [FFmpeg-user] FFmpeg

[Gyan Doshi]

On 2024-05-17 10:12 am, Keith Reno wrote:

I have ffmpeg-7.0-full_build downloaded on my computer.  When I select ffmpeg 
the screen flashes but nothing loads.  Do I have the correct version or am I 
doing something wrong?


Read 
https://video.stackexchange.com/questions/20495/how-do-i-set-up-and-use-ffmpeg-in-windows/


Regards,
Gyan

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[FFmpeg-user] FFmpeg

[Keith Reno]

I have ffmpeg-7.0-full_build downloaded on my computer.  When I select ffmpeg 
the screen flashes but nothing loads.  Do I have the correct version or am I 
doing something wrong?

Keith

Sent from Mail for Windows

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[FFmpeg-user] FFMPEG concatenating a transcode as 59.88 fps instead of 60 fps

[Ryan Quinlan]

Hello all,

I am transcoding an image sequence to a 10 frame 60fps .mov files using the
first command below. I am then concatenating the exported videos using the
second command below.

*Transcode*

ffmpeg -y -r 60 -probesize 13M -start_number 1001 -i
'input_video.%04d.jpeg' -c:v libx265 -pix_fmt yuv420p10le -crf 28
-preset slower -tag:v hvc1 -vf lut3d='output_lut.cube' -r 60 -frames:v
10 -movflags faststart output_video_0001.mov

*Concatenate*

ffmpeg -y -f concat -safe 0 -i input_concat_list.txt -c copy output_concat.mov

I expected the final file from the concatenated videos to be 60fps. I am
using the -r flag twice, so it sets the input frame rate for the image
sequence and then making sure the output is 60fps as well. After the
transcode the resulting chunks are each 60fps. After the concatenation
though, when checking the frame rate in VLC, instead of the desired 60fps
VLC is showing the resulting video is 59.979960. I tried using the frame
rate flag when concatenating, but that didn't help.
Thank you for any assistance.

Ryan
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Re: [FFmpeg-user] [FFmpeg] getting errors while running ffmpeg locally.

[B-2014 Ariyan Kashyap]

Hello Reindl,
Thank you for reverting back
Following are the details you have asked
package version : 6.1.1
Operating System : Windows 10

I will install the nasm and yasm packages and revert back to you after the
installation.
Thanks again.

On Tue, Apr 16, 2024 at 7:18 PM Reindl Harald 
wrote:

>
>
> Am 16.04.24 um 15:43 schrieb B-2014 Ariyan Kashyap:
> > Getting this err while running ./configure --enable-shared
> >
> > ./configure --enable-shared
> > nasm/yasm not found or too old. Use --disable-x86asm for a crippled
> build.
>
> so why don't you just install "nasm" or "yasm" or at least tell us your
> operating system and versions
>
> i can't think of a clearer message - package missing or too old version
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Re: [FFmpeg-user] [FFmpeg] getting errors while running ffmpeg locally.

[Reindl Harald]

Am 16.04.24 um 15:43 schrieb B-2014 Ariyan Kashyap:

Getting this err while running ./configure --enable-shared

./configure --enable-shared
nasm/yasm not found or too old. Use --disable-x86asm for a crippled build.


so why don't you just install "nasm" or "yasm" or at least tell us your 
operating system and versions


i can't think of a clearer message - package missing or too old version
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[FFmpeg-user] [FFmpeg] getting errors while running ffmpeg locally.

[B-2014 Ariyan Kashyap]

Getting this err while running ./configure --enable-shared

./configure --enable-shared
nasm/yasm not found or too old. Use --disable-x86asm for a crippled build.

If you think configure made a mistake, make sure you are using the latest
version from Git.  If the latest version fails, report the problem to the
ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on irc.libera.chat.
Include the log file "ffbuild/config.log" produced by configure as this
will help
solve the problem.
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[FFmpeg-user] ffmpeg doesn't generate a ISO/IEC 14496-17 (MPEG-4 text) track when ingesting WebVTT subtitles to produce MPEGTS

[Daniel Andres Pelaez Lopez]

Hi everyone,

We are trying to create a mpegts with a ISO/IEC 14496-17 (MPEG-4 text)
subtitles track, using WebVTT, but seems like ffmpeg creates a ISO
13818-1 PES private data instead. The following is the ffmpeg command:

ffmpeg -i subtitle.vtt -c:s mov_text -f mpegts output3GPP.ts

And the following is the output of the ffprobe for that file:

Input #0, mpegts, from 'output3GPP.ts':
  Duration: 00:00:19.00, start: 1.40, bitrate: 1 kb/s
  Program 1
Metadata:
  service_name: Service01
  service_provider: FFmpeg
  Stream #0:0[0x100]: Data: bin_data ([6][0][0][0] / 0x0006)
Unsupported codec with id 98314 for input stream 0

However, if we generate a mp4, it works, the following is the command:

ffmpeg -i subtitle.vtt -c:s mov_text -f mp4 output3GPP.ts

And the following is the output of the ffprobe for that file:

Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output3GPP.ts':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2mp41
encoder : Lavf57.83.100
  Duration: 00:00:20.00, start: 0.00, bitrate: 0 kb/s
  Stream #0:0[0x1](und): Subtitle: mov_text (tx3g / 0x67337874), 0
kb/s (default)
Metadata:
  handler_name: SubtitleHandler

Any reason why both commands behave differently? is mpegts not
supporting ISO/IEC 14496-17 (MPEG-4 text)?

Note: attached is the subtitle.vtt file.

Regards


-- 
Daniel Andrés Pelaez López
WEBVTT

00:00.000 --> 00:01.000
Subtitle 1

00:01.000 --> 00:02.000
Subtitle 2

00:02.000 --> 00:03.000
Subtitle 3

00:03.000 --> 00:04.000
Subtitle 4

00:04.000 --> 00:05.000
Subtitle 5

00:05.000 --> 00:06.000
Subtitle 6

00:06.000 --> 00:07.000
Subtitle 7

00:07.000 --> 00:08.000
Subtitle 8

00:08.000 --> 00:09.000
Subtitle 9

00:09.000 --> 00:10.000
Subtitle 10

00:10.000 --> 00:11.000
Subtitle 11

00:11.000 --> 00:12.000
Subtitle 12

00:12.000 --> 00:13.000
Subtitle 13

00:13.000 --> 00:14.000
Subtitle 14

00:14.000 --> 00:15.000
Subtitle 15

00:15.000 --> 00:16.000
Subtitle 16

00:16.000 --> 00:17.000
Subtitle 17

00:17.000 --> 00:18.000
Subtitle 18

00:18.000 --> 00:19.000
Subtitle 19

00:19.000 --> 00:20.000
Subtitle 20

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Re: [FFmpeg-user] FFmpeg Architecture

[Reindl Harald]

Am 14.02.24 um 08:51 schrieb Paul B Mahol:

Because order of arguments in FFmpeg matters.


and why didn't you respond exactly that days ago instead your foolish 
answer which can even be interpreted as you don't know that the order 
matters and pretend the OP is facing ghosts


 Weitergeleitete Nachricht 
Betreff: Re: [FFmpeg-user] FFmpeg Architecture
Datum: Tue, 13 Feb 2024 19:39:54 +0100
Von: Paul B Mahol 
Antwort an: FFmpeg user questions 
An: FFmpeg user questions 

On Tue, Feb 13, 2024 at 7:21 PM Mark Filipak 
wrote:

> On 13/02/2024 03.24, Paul B Mahol wrote:
> > On Tue, Feb 13, 2024 at 12:37 AM Carl Zwanzig  wrote:
>
> Paul, Carl, I assume you are both familiar with the source code. I'm not.
>
> Fact: '-ss'-before-'-i' and '-i'-before-'-ss' _do_ give different 
results.

> I suggested that the
> difference might result from the way that FFmpeg is designed.
>

Not fact, your logic and math skills are really poor, reconsider 
additional real educational summer camp.

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Re: [FFmpeg-user] FFmpeg Architecture

[Erik Dobberkau]

> Why you insist on 'should' is question for you, I doubt we will ever get
> the answer.
>

My impression is he's looking for a way to use/turn a swiss army knife
as/into a scalpel, if this analogy makes sense.
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Re: [FFmpeg-user] FFmpeg Architecture

[Paul B Mahol]

On Wed, Feb 14, 2024 at 3:11 AM Mark Filipak 
wrote:

> On 13/02/2024 20.27, amindfv--- via ffmpeg-user wrote:
> >>> params *before* "-i" are *input params*
> >>> params *after* "-i" are *output params*
> >>>
> >>> that has *always* been the case at least for 15 years and so it's
> >>> logical that you can't place a param at a random position by common
> >>> sense and expect the same result
> >>
> >> Thank you, Harald. I knew that but your gentle reminder is welcome. It
> has
> >> helped me to better see the problem.
> >
> > This is documented in the first 2 sections here:
> https://trac.ffmpeg.org/wiki/Seeking
>
> Yes, Tom, and it essentially says they are very different in execution
> time but identical in result,
> which they're not, not even close.
>
> Also, '-i'-before-'-ss' is ignoring '-copyts'.
> Also, 'too-early' video packets are being output instead of cut.
> Also, 'too-early' audio packets are being output instead of cut.
> Also, the first audio packet that is erroneously passed is much smaller
> and has a different CRC than
> it should have even though this is a remuxing operation.
> The evidence of bugs throughout is overwhelming.
>
> Contrary to what's written, '-i this.m2ts -ss 20 -to 60' outputs 40
> seconds, not 60.
>
> Shall I go on?
>
> My question once again, in this thread's first post:
>
> Why are these different?
> ffmpeg -ss 20.061708333 -copyts -i %SOURCE% -map 0 -c copy -sn -dn
> -muxdelay 0 %TARGET%
> ffmpeg -copyts -i %SOURCE% -map 0 -ss 20.061708333 -c copy -sn -dn
> -muxdelay 0 %TARGET%
>
> Note that "20.061708333" is a time, not a PTS.
>
> Why are they hundreds of frames different?
>

Because order of arguments in FFmpeg matters.


>
> We are over 35 posts in this thread (not counting 2 other threads of
> related stuff) and no one has
> even said whether they should or should not be the same.
>

I wrote it already that your requests are invalid.

Your questions is assuming only 1 possible outcome of only 2 possible cases.

I wrote already, it might or might not be the same.


Why you insist on 'should' is question for you, I doubt we will ever get
the answer.



>
>
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Re: [FFmpeg-user] FFmpeg Architecture

[Mark Filipak]

On 13/02/2024 20.27, amindfv--- via ffmpeg-user wrote:

params *before* "-i" are *input params*
params *after* "-i" are *output params*

that has *always* been the case at least for 15 years and so it's
logical that you can't place a param at a random position by common
sense and expect the same result


Thank you, Harald. I knew that but your gentle reminder is welcome. It has
helped me to better see the problem.


This is documented in the first 2 sections here: 
https://trac.ffmpeg.org/wiki/Seeking


Yes, Tom, and it essentially says they are very different in execution time but identical in result, 
which they're not, not even close.


Also, '-i'-before-'-ss' is ignoring '-copyts'.
Also, 'too-early' video packets are being output instead of cut.
Also, 'too-early' audio packets are being output instead of cut.
Also, the first audio packet that is erroneously passed is much smaller and has a different CRC than 
it should have even though this is a remuxing operation.

The evidence of bugs throughout is overwhelming.

Contrary to what's written, '-i this.m2ts -ss 20 -to 60' outputs 40 seconds, 
not 60.

Shall I go on?

My question once again, in this thread's first post:

Why are these different?
ffmpeg -ss 20.061708333 -copyts -i %SOURCE% -map 0 -c copy -sn -dn -muxdelay 0 
%TARGET%
ffmpeg -copyts -i %SOURCE% -map 0 -ss 20.061708333 -c copy -sn -dn -muxdelay 0 
%TARGET%

Note that "20.061708333" is a time, not a PTS.

Why are they hundreds of frames different?

We are over 35 posts in this thread (not counting 2 other threads of related stuff) and no one has 
even said whether they should or should not be the same.



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Re: [FFmpeg-user] FFmpeg Architecture

[amindfv--- via ffmpeg-user]

> > params *before* "-i" are *input params*
> > params *after* "-i" are *output params*
> > 
> > that has *always* been the case at least for 15 years and so it's
> > logical that you can't place a param at a random position by common
> > sense and expect the same result
> 
> Thank you, Harald. I knew that but your gentle reminder is welcome. It has
> helped me to better see the problem.

This is documented in the first 2 sections here: 
https://trac.ffmpeg.org/wiki/Seeking

Tom
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Re: [FFmpeg-user] FFmpeg Architecture

[Mark Filipak]

On 13/02/2024 14.13, Reindl Harald wrote:


Am 13.02.24 um 19:39 schrieb Paul B Mahol:

On Tue, Feb 13, 2024 at 7:21 PM Mark Filipak 
wrote:


On 13/02/2024 03.24, Paul B Mahol wrote:

On Tue, Feb 13, 2024 at 12:37 AM Carl Zwanzig  wrote:


Paul, Carl, I assume you are both familiar with the source code. I'm not.

Fact: '-ss'-before-'-i' and '-i'-before-'-ss' _do_ give different results.
I suggested that the
difference might result from the way that FFmpeg is designed.



Not fact, your logic and math skills are really poor, reconsider additional
real educational summer camp.


params *before* "-i" are *input params*
params *after* "-i" are *output params*

that has *always* been the case at least for 15 years and so it's logical that you can't place a 
param at a random position by common sense and expect the same result


Thank you, Harald. I knew that but your gentle reminder is welcome. It has helped me to better see 
the problem.


'-i'-before-'-ss' is not honoring '-copyts' and possibly '-muxdelay 0' so the output's PTSs do not 
match what '-ss' is asking for because '-ss' is citing input PTSs. Is that what you're 'saying'?


--Mark.


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Re: [FFmpeg-user] FFmpeg Architecture

[Reindl Harald]

Am 13.02.24 um 19:39 schrieb Paul B Mahol:

On Tue, Feb 13, 2024 at 7:21 PM Mark Filipak 
wrote:


On 13/02/2024 03.24, Paul B Mahol wrote:

On Tue, Feb 13, 2024 at 12:37 AM Carl Zwanzig  wrote:


Paul, Carl, I assume you are both familiar with the source code. I'm not.

Fact: '-ss'-before-'-i' and '-i'-before-'-ss' _do_ give different results.
I suggested that the
difference might result from the way that FFmpeg is designed.



Not fact, your logic and math skills are really poor, reconsider additional
real educational summer camp.


params *before* "-i" are *input params*
params *after* "-i" are *output params*

that has *always* been the case at least for 15 years and so it's 
logical that you can't place a param at a random position by common 
sense and expect the same result

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Re: [FFmpeg-user] FFmpeg Architecture

[Paul B Mahol]

On Tue, Feb 13, 2024 at 7:21 PM Mark Filipak 
wrote:

> On 13/02/2024 03.24, Paul B Mahol wrote:
> > On Tue, Feb 13, 2024 at 12:37 AM Carl Zwanzig  wrote:
>
> Paul, Carl, I assume you are both familiar with the source code. I'm not.
>
> Fact: '-ss'-before-'-i' and '-i'-before-'-ss' _do_ give different results.
> I suggested that the
> difference might result from the way that FFmpeg is designed.
>

Not fact, your logic and math skills are really poor, reconsider additional
real educational summer camp.


>
> I mentioned dynamic versus static architectures. When thinking about
> dynamic versus static think
> "build up/tear down" versus "call". "Build up/tear down" versus "call" has
> to do with implementation
> -- how FFmpeg runs jobs -- not whether the job is streaming or real-time.
> In H.222, ISO & ITU imply
> (to me) that they had "build up/tear down" in mind when they created the
> system model they created.
>
> The question is this: Should '-ss'-before-'-i' and '-i'-before-'-ss' give
> different results? They
> _do_, but _should_ they?
>

No, it might or might not give different results.


>
> Set aside for the moment that '-ss'-before-'-i' and '-i'-before-'-ss' make
> differing cuts (at
> differing PTSs) that are hundreds of frames different.
> Set aside for the moment that '-ss'-before-'-i' honors '-muxdelay 0' but
> '-i'-before-'-ss' doesn't.
> Set aside for the moment that both of them cut the 'earliest' audio
> packets but both leave some
> 'too-early' audio packets in their results.
> Set aside for the moment that both of them include video packets that have
> 'too-early' PTSs that
> should have been cut.
>
> None of that matters for now.
>
> I can't submit a bug ticket unless I know whether '-ss'-before-'-i' and
> '-i'-before-'-ss' should
> give the same results. After that is established and I submit a ticket,
> assuming the answer is that
> they should be the same, then all the other details will be hashed out and
> all the fighting will
> occur in ... github? Is it github? I think so.
>

There is nothing to fix, bug is not here.


>
> Please help with this if you can.
>
> --Mark.
>
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Re: [FFmpeg-user] FFmpeg Architecture

[Mark Filipak]

On 13/02/2024 03.24, Paul B Mahol wrote:

On Tue, Feb 13, 2024 at 12:37 AM Carl Zwanzig  wrote:


Paul, Carl, I assume you are both familiar with the source code. I'm not.

Fact: '-ss'-before-'-i' and '-i'-before-'-ss' _do_ give different results. I suggested that the 
difference might result from the way that FFmpeg is designed.


I mentioned dynamic versus static architectures. When thinking about dynamic versus static think 
"build up/tear down" versus "call". "Build up/tear down" versus "call" has to do with implementation 
-- how FFmpeg runs jobs -- not whether the job is streaming or real-time. In H.222, ISO & ITU imply 
(to me) that they had "build up/tear down" in mind when they created the system model they created.


The question is this: Should '-ss'-before-'-i' and '-i'-before-'-ss' give different results? They 
_do_, but _should_ they?


Set aside for the moment that '-ss'-before-'-i' and '-i'-before-'-ss' make differing cuts (at 
differing PTSs) that are hundreds of frames different.

Set aside for the moment that '-ss'-before-'-i' honors '-muxdelay 0' but 
'-i'-before-'-ss' doesn't.
Set aside for the moment that both of them cut the 'earliest' audio packets but both leave some 
'too-early' audio packets in their results.
Set aside for the moment that both of them include video packets that have 'too-early' PTSs that 
should have been cut.


None of that matters for now.

I can't submit a bug ticket unless I know whether '-ss'-before-'-i' and '-i'-before-'-ss' should 
give the same results. After that is established and I submit a ticket, assuming the answer is that 
they should be the same, then all the other details will be hashed out and all the fighting will 
occur in ... github? Is it github? I think so.


Please help with this if you can.

--Mark.

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Re: [FFmpeg-user] FFmpeg Architecture

[Carl Zwanzig]

On 2/13/2024 8:37 AM, Carl Zwanzig wrote:

"read time"


reaL time

z!

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Re: [FFmpeg-user] FFmpeg Architecture

[Carl Zwanzig]

On 2/13/2024 12:24 AM, Paul B Mahol wrote:

Even if implemented properly it will not make real-time not working.
FFmpeg does not have real variable filter chains. Everything is static,
there are only filters that can be auto inserted once in a while.


Granted, AFAICT much of how ffmpeg is used is very much not "read time" as 
commonly understood in the computing/process-control world. it would be 
interesting if the filter chain could be modified conditionally, on a 
frame-by-frame basis, but that would definitely require some work. But a lot 
of this discussion hinges around a very nebulous idea of what any 
"scripting" would do, and without a better understanding of that it's hard 
to have meaningful discussion.


And. much of what might be considered "scripting" looks what others call 
using an "edit decision list" (EDL), and there are multiple ffmpeg-based 
packages that already implement those (e.g. OpenShot, Shotcut, Cinelarra 
etc) and with a nice UI :).


Later,

z!

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Re: [FFmpeg-user] FFmpeg Architecture

[Paul B Mahol]

On Tue, Feb 13, 2024 at 12:37 AM Carl Zwanzig  wrote:

> On 2/12/2024 2:59 PM, Paul B Mahol wrote:
> > I doubt anything above applies to FFmpeg, also I doubt it makes
> difference
> > for real-time streaming.
>
> Hmm, actually it looks like a lot of it could apply if implemented
> properly
> (there are a fair few video programs which implement some form of
> scripting,
> OBS for one). And from the right angle, the variable filter chains could
> be
> considered "scripting", too.
>

Even if implemented properly it will not make real-time not working.
FFmpeg does not have real variable filter chains. Everything is static,
there are only filters that can be auto inserted once in a while.


>
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Re: [FFmpeg-user] FFmpeg Architecture

[Reindl Harald]

Am 13.02.24 um 00:31 schrieb Bouke / Videotoolshed:

Go away, or at least shut up


after dozens of useless posts why don't you?

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Re: [FFmpeg-user] FFmpeg Architecture

[Carl Zwanzig]

On 2/12/2024 3:31 PM, Bouke / Videotoolshed wrote:

Go away, or at least shut up.


I encourage you do do that if you can't add to the conversation.

z!
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Re: [FFmpeg-user] FFmpeg Architecture

[Carl Zwanzig]

On 2/12/2024 2:59 PM, Paul B Mahol wrote:

I doubt anything above applies to FFmpeg, also I doubt it makes difference
for real-time streaming.


Hmm, actually it looks like a lot of it could apply if implemented properly 
(there are a fair few video programs which implement some form of scripting, 
OBS for one). And from the right angle, the variable filter chains could be 
considered "scripting", too.


z!
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Re: [FFmpeg-user] FFmpeg Architecture (was: Codec copy with -ss before -i and after -i is not documented)

[Bouke / Videotoolshed]

Even though I used the word "or" in my post, the use of any of the 
architectural techniques can be mixed.

On the subject of you being a troll, you wrote that you posted no more than 6 
posts a year.

That turns out to be ‘a day’

Go away, or at least shut up.


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Re: [FFmpeg-user] FFmpeg Architecture (was: Codec copy with -ss before -i and after -i is not documented)

[Paul B Mahol]

On Mon, Feb 12, 2024 at 11:40 PM Mark Filipak 
wrote:

> On 12/02/2024 11.30, Paul B Mahol wrote:
> > On Mon, Feb 12, 2024 at 9:48 AM Mark Filipak  >
> > wrote:
> >> Does FFmpeg do any internal scripting or dynamic code building/linking?
> >
> > Please define what you mean by these concepts.
>
> Happy to, Paul.
>
> Internal scripting is when a task is built using a scripting language (or
> pseudo language) that is
> then run by submitting it to a built-in interpreter -- much as you script
> a task in a terminal --
> except that the interpreter and the script aren't made or seen by the
> user. I think that in video,
> such an internal script is sometimes called a workflow but I'm unsure of
> that. After the script has
> been run, it's usually discarded.
>
> Dynamic, as opposed to static, means that execution changes based on
> run-time conditions that
> usually are not explicitly controlled by the user. It appears that FFmpeg
> currently does not use any
> dynamic code. Conditional logic alone does not make code dynamic.
>
> Code building is when existing code blocks, such as library functions, are
> copied and linked. The
> behavior is very much like internal scripting but with faster execution.
> Code linking accomplishes
> the same result but via code blocks that accept parameters and are
> re-entrant. Dynamic code
> building/linking means that the copying/linking happens at run-time based
> on run-time conditions
> that are not explicitly controlled by the user.
>
> Because they are run-time code structures, the dynamic stuff isn't suited
> to streaming because
> run-time in streaming means real-time. Dynamic is not well suited to
> real-time processing. Running
> an interpreter of any kind is not well suited to real-time, either.
>

> Even though I used the word "or" in my post, the use of any of the
> architectural techniques can be
> mixed.
>

I doubt anything above applies to FFmpeg, also I doubt it makes difference
for real-time streaming.


>
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Re: [FFmpeg-user] FFmpeg Architecture (was: Codec copy with -ss before -i and after -i is not documented)

[Mark Filipak]

On 12/02/2024 11.30, Paul B Mahol wrote:

On Mon, Feb 12, 2024 at 9:48 AM Mark Filipak 
wrote:

Does FFmpeg do any internal scripting or dynamic code building/linking?


Please define what you mean by these concepts.


Happy to, Paul.

Internal scripting is when a task is built using a scripting language (or pseudo language) that is 
then run by submitting it to a built-in interpreter -- much as you script a task in a terminal -- 
except that the interpreter and the script aren't made or seen by the user. I think that in video, 
such an internal script is sometimes called a workflow but I'm unsure of that. After the script has 
been run, it's usually discarded.


Dynamic, as opposed to static, means that execution changes based on run-time conditions that 
usually are not explicitly controlled by the user. It appears that FFmpeg currently does not use any 
dynamic code. Conditional logic alone does not make code dynamic.


Code building is when existing code blocks, such as library functions, are copied and linked. The 
behavior is very much like internal scripting but with faster execution. Code linking accomplishes 
the same result but via code blocks that accept parameters and are re-entrant. Dynamic code 
building/linking means that the copying/linking happens at run-time based on run-time conditions 
that are not explicitly controlled by the user.


Because they are run-time code structures, the dynamic stuff isn't suited to streaming because 
run-time in streaming means real-time. Dynamic is not well suited to real-time processing. Running 
an interpreter of any kind is not well suited to real-time, either.


Even though I used the word "or" in my post, the use of any of the architectural techniques can be 
mixed.


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Re: [FFmpeg-user] FFmpeg Architecture (was: Codec copy with -ss before -i and after -i is not documented)

[Paul B Mahol]

On Mon, Feb 12, 2024 at 9:48 AM Mark Filipak 
wrote:

> On 12/02/2024 03.21, Paul B Mahol wrote:
> > Mark wants single command example to work 100% correct in 100% situations
> > and that output is 100% correct 100% of time.
>
> Yes. Of course.
>
> > That is possible only for very trivial containers only currently. And
> > because libs are very low level it can not be done at all with current
> > state of code.
>
> Now I think we're making some progress. Tell me, Paul, does FFmpeg do
> anything beyond what its
> encoders do? I'm pretty sure the answer is "yes" for '-vf' and '-bsf'. Are
> there any others?
>
> Does FFmpeg do any internal scripting or dynamic code building/linking?
>

Please define what you mean by these concepts.


>
> --Mark.
>
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[FFmpeg-user] FFmpeg Architecture (was: Codec copy with -ss before -i and after -i is not documented)

[Mark Filipak]

On 12/02/2024 03.21, Paul B Mahol wrote:

Mark wants single command example to work 100% correct in 100% situations
and that output is 100% correct 100% of time.


Yes. Of course.


That is possible only for very trivial containers only currently. And
because libs are very low level it can not be done at all with current
state of code.


Now I think we're making some progress. Tell me, Paul, does FFmpeg do anything beyond what its 
encoders do? I'm pretty sure the answer is "yes" for '-vf' and '-bsf'. Are there any others?


Does FFmpeg do any internal scripting or dynamic code building/linking?

--Mark.

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Re: [FFmpeg-user] FFmpeg interrupted (SIGSTOP signal) but pipewire shows its node is still running

[gnusuario--- via ffmpeg-user]

On Mon, 05 Feb 2024 08:34:09 GMT, Nicolas George

geo...@nsup.org wrote:

gnusuario--- via ffmpeg-user (12024-02-05): > I want to know if it is possible 
to stop recording sound while the > process is paused (stopped in signal terms).

FFmpeg cannot do ANYTHING after receiving a SIGSTOP. So you would need either 
sending TSTP and see if it behaves better, or take it to the authors of 
pipewire.

Regards,

-- Nicolas George ___ ffmpeg-user 
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I tried with SIGTSTP, no change or even worse. If ffmpeg is launched without a 
terminal emulator not only does audio continue being recorded, but video too. 
Ps output shows state Sl, so it's running. Apparently SIGTSTP is not caught.

Thanks.
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Re: [FFmpeg-user] FFmpeg interrupted (SIGSTOP signal) but pipewire shows its node is still running

[Nicolas George]

gnusuario--- via ffmpeg-user (12024-02-05):
> I want to know if it is possible to stop recording sound while the
> process is paused (stopped in signal terms).  

FFmpeg cannot do ANYTHING after receiving a SIGSTOP. So you would need
either sending TSTP and see if it behaves better, or take it to the
authors of pipewire.

Regards,

-- 
  Nicolas George
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Re: [FFmpeg-user] FFmpeg interrupted (SIGSTOP signal) but pipewire shows its node is still running

[gnusuario--- via ffmpeg-user]

To be more precise, the node in pw-top is named after the library FFmpeg uses: 
Lavf60.16.100. In top all threads of ffmpeg are suspended (T).

On Mon, 05 Feb 2024 00:33:20 GMT, gnusuario--- via ffmpeg-user

ffmpeg-user@ffmpeg.org wrote:

Forgot to mention that the FFmpeg version is the ome from Packman repositories. 
6.1.1.

On Mon, 05 Feb 2024 00:29:13 GMT, gnusuario--- via ffmpeg-user

ffmpeg-user@ffmpeg.org wrote:

I'm using FFmpeg to save screencasts to disk. I start FFmpeg and pause, 
continue or stop the screencast by sending signals to the process. I found out 
that in recorded screencasts with pauses the audio does not pause and continues 
as if FFmpeg was actively running. Because of that, after resuming the 
recording the audio is out of sync. The audio between the interruption and the 
resuming is included in the recording. While interrupted ffmpeg's status is Tt 
as printed by ps. On the other hand pw-top shows ffmpeg's node with status R 
(running). My OS is openSUSE Tumbleweed 20240202. The command I'm executing is 
as follows: ffmpeg -device /device/dri/card1 -framerate 50 -f kmsgrab -i - -f 
pulse -i $(pactl get-default-sink).monitor -filter_complex "[0:v] 
setpts=N/(FR*TB),hwmap=derive_device=vaapi,fps=fps=50,scale_vaapi=format=nv12;[1:a]
 asetpts=N/(SR*TB);" -ac 2 -c:a libopus -b:a 128k -c:v hevc_vaapi -bf:v 8 
-rc_mode:v CQP -profile:v main -tier:v main -level:v 4.0 -global_qualit y
  :v 24 -qmin:v 20 -y output.mkv & Then I send to ffmpeg signals as needed, 
with kill. kill -SIGINT ${pid}, kill -SIGSTOP ${pid}, kill -SIGCONT ${pid}. I 
want to know if it is possible to stop recording sound while the process is 
paused (stopped in signal terms). 
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Re: [FFmpeg-user] FFmpeg interrupted (SIGSTOP signal) but pipewire shows its node is still running

[gnusuario--- via ffmpeg-user]

Forgot to mention that the FFmpeg version is the ome from Packman repositories. 
6.1.1.

On Mon, 05 Feb 2024 00:29:13 GMT, gnusuario--- via ffmpeg-user

ffmpeg-user@ffmpeg.org wrote:

I'm using FFmpeg to save screencasts to disk. I start FFmpeg and pause, 
continue or stop the screencast by sending signals to the process. I found out 
that in recorded screencasts with pauses the audio does not pause and continues 
as if FFmpeg was actively running. Because of that, after resuming the 
recording the audio is out of sync. The audio between the interruption and the 
resuming is included in the recording. While interrupted ffmpeg's status is Tt 
as printed by ps. On the other hand pw-top shows ffmpeg's node with status R 
(running). My OS is openSUSE Tumbleweed 20240202. The command I'm executing is 
as follows: ffmpeg -device /device/dri/card1 -framerate 50 -f kmsgrab -i - -f 
pulse -i $(pactl get-default-sink).monitor -filter_complex "[0:v] 
setpts=N/(FR*TB),hwmap=derive_device=vaapi,fps=fps=50,scale_vaapi=format=nv12;[1:a]
 asetpts=N/(SR*TB);" -ac 2 -c:a libopus -b:a 128k -c:v hevc_vaapi -bf:v 8 
-rc_mode:v CQP -profile:v main -tier:v main -level:v 4.0 -global_qualit y
 :v 24 -qmin:v 20 -y output.mkv & Then I send to ffmpeg signals as needed, with 
kill. kill -SIGINT ${pid}, kill -SIGSTOP ${pid}, kill -SIGCONT ${pid}. I want 
to know if it is possible to stop recording sound while the process is paused 
(stopped in signal terms). ___ 
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[FFmpeg-user] FFmpeg interrupted (SIGSTOP signal) but pipewire shows its node is still running

[gnusuario--- via ffmpeg-user]

I'm using FFmpeg to save screencasts to disk. I start FFmpeg and pause, 
continue or stop the screencast by sending signals to the process. I found out 
that in recorded screencasts with pauses the audio does not pause and continues 
as if FFmpeg was actively running. Because of that, after resuming the 
recording the audio is out of sync. The audio between the interruption and the 
resuming is included in the recording. While interrupted ffmpeg's 
status is Tt as printed by ps. On the other hand pw-top shows ffmpeg's node 
with status R (running). My OS is openSUSE Tumbleweed 20240202.  The 
command I'm executing is as follows: ffmpeg -device /device/dri/card1 
-framerate 50 -f kmsgrab -i - -f pulse -i $(pactl get-default-sink).monitor 
-filter_complex "[0:v] 
setpts=N/(FR*TB),hwmap=derive_device=vaapi,fps=fps=50,scale_vaapi=format=nv12;[1:a]
 asetpts=N/(SR*TB);" -ac 2 -c:a libopus -b:a 128k -c:v hevc_vaapi -bf:v 8 
-rc_mode:v CQP -profile:v main -tier:v main -level:v 4.0 -global_qualit
 y:v 24 -qmin:v 20 -y output.mkv &  Then I send to ffmpeg signals 
as needed, with kill. kill -SIGINT ${pid}, kill -SIGSTOP ${pid}, kill -SIGCONT 
${pid}.I want to know if it is possible to stop recording sound 
while the process is paused (stopped in signal terms). 
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[FFmpeg-user] ffmpeg destroying the video stream when copying from m2ts to m2ts

[Binarus]

Dear all,

I have a m2ts file that contains one video track and 14 other audio and 
subtitle tracks. I'd like to copy the video track into a new m2ts file. 
This basically works, but ffmpeg is doing something bad to the video 
track while copying.


The new file plays in VLC and MPC-HC, but the video track is massively 
disturbed. For some seconds, it plays normally, for the next seconds, it 
shows just random color mess, then it plays normally again, and so on.


When converting to mkv instead, the new file plays correctly.

This is the ffmpeg version:

ffmpeg version 2024-01-14-git-34a47b97de-full_build-www.gyan.dev 
Copyright (c) 2000-2024 the FFmpeg developers

  built with gcc 12.2.0 (Rev10, Built by MSYS2 project)

This is the command line to extract the video stream from the m2ts file 
into a new m2ts file:


ffmpeg -i in.m2ts -map 0:0 -codec copy out.m2ts

I also have successively added -copyts, -muxpreload 0, -muxdelay 0, 
-probesize 100G and -analyzeduration 100G to that command line at the 
correct places, and also have tried combinations thereof, but it didn't 
change anything.


These are the streams in the input file:

  Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448), 
yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 
tbr, 90k tbn
  Stream #0:1[0x1100]: Audio: dts (DTS-HD MA) ([134][0][0][0] / 
0x0086), 48000 Hz, 5.1(side), s16p 

  Stream #0:2[0x1101]: Audio: dts (DTS-HD MA) ([134][0][0][0] / 
0x0086), 48000 Hz, 5.1(side), s16p 

  Stream #0:3[0x1102]: Audio: dts (DTS-HD MA) ([134][0][0][0] / 
0x0086), 48000 Hz, 5.1(side), s16p 

  Stream #0:4[0x1103]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 
stereo, fltp, 192 kb/s 

  Stream #0:5[0x1104]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 
5.1(side), fltp, 640 kb/s 

  Stream #0:6[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090)
  Stream #0:7[0x1201]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090)
  Stream #0:8[0x1202]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090)
  Stream #0:9[0x1203]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090)
  Stream #0:10[0x1204]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090)
  Stream #0:11[0x1205]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090)
  Stream #0:12[0x1206]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090)
  Stream #0:13[0x1207]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090)
  Stream #0:14[0x1208]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 
0x0090), 1920x1080


I'd be happy to provide the file for further analysis. However, I can't 
make the file public. The file is about 300 MB; I can upload it 
somewhere and provide the link. If somebody is interested, I'd like to 
discuss the further procedure via personal email; please feel free to 
drop me a note.


Best regards, and thank you very much,

Binarus

P.S. I don't know yet whether the problem is specific only to the input 
file in question.

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Re: [FFmpeg-user] [FFmpeg-devel] FFmpeg's HLS muxer's deprecation of the hls_wrap option

[Steven Liu]

Dennis Mungai 于2024年1月22日 周一18:20写道：

> On Mon, 22 Jan 2024 at 13:10, Steven Liu  wrote:
>
> > Dennis Mungai  于2024年1月21日周日 00:42写道：
> >
> > > >
> > > Perfect, and thanks for considering a revert of this deprecated
> feature.
> >
> > Hi Dennis,
> >
> > Need more talk, Why don't try to use hls_flags
> > +delete_segments+append_list. ?
> >
> >
> > Thanks
> > Steven
> >
>
> As stated earlier, even with these flags, we cannot replicate the advantage

No I don‘t think that is advantage, because hls_wrap can cover the old
segments with same segment file name.
And the deprecated warning has been there more than five years I think.

>
> of the deprecated -hls_wrap whereby the segment names remain constant.
> append_list only omits the #EXT-EXT-ENDLIST tag.




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Re: [FFmpeg-user] [FFmpeg-devel] FFmpeg's HLS muxer's deprecation of the hls_wrap option

[Dennis Mungai]

On Mon, 22 Jan 2024 at 13:10, Steven Liu  wrote:

> Dennis Mungai  于2024年1月21日周日 00:42写道：
>
> > >
> > Perfect, and thanks for considering a revert of this deprecated feature.
>
> Hi Dennis,
>
> Need more talk, Why don't try to use hls_flags
> +delete_segments+append_list. ?
>
>
> Thanks
> Steven
>

As stated earlier, even with these flags, we cannot replicate the advantage
of the deprecated -hls_wrap whereby the segment names remain constant.
append_list only omits the #EXT-EXT-ENDLIST tag.
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Re: [FFmpeg-user] [FFmpeg-devel] FFmpeg's HLS muxer's deprecation of the hls_wrap option

[Steven Liu]

Dennis Mungai  于2024年1月21日周日 00:42写道：

> >
> Perfect, and thanks for considering a revert of this deprecated feature.

Hi Dennis,

Need more talk, Why don't try to use hls_flags +delete_segments+append_list. ?


Thanks
Steven
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Re: [FFmpeg-user] [FFmpeg-devel] FFmpeg's HLS muxer's deprecation of the hls_wrap option

[Dennis Mungai]

On Sat, 20 Jan 2024 at 18:42, Steven Liu  wrote:

> Dennis Mungai 于2024年1月20日 周六23:15写道：
>
> > On Sat, 20 Jan 2024, 6:09 pm Steven Liu, 
> wrote:
> >
> > > Dennis Mungai 于2024年1月20日 周六21:02写道：
> > >
> > > > Hello,
> > > >
> > > > Is there a valid technical reason as to why FFmpeg's HLS muxer
> dropped
> > > the
> > > > hls_wrap option?
> > >
> > >
> > > eg. three player playing the list and every fragment less or equal 1
> > second
> > > ,
> > > 1. Player 1 from fragment1, fragment2
> > > 2. Now Player 2 from fragment 1, fragment 2,
> > > 3. Now Player 3 from fragment 1, Fragment2 and Player 1 playing
> fragment
> > 3
> > > 4. Network transport with player 3 get loss packet
> > > 5. Player 1 playing new fragment 1, new fragment 2
> > > 6. Player 2 playing fragment 3
> > > 7. Player 3 blocking always because the fragments are flashing too
> fast.
> > >
> > > So the commit message said it is not friendly to downstream users.
> > >
> > > I have no more better way to fix it with ffmpeg, but I think that
> commit
> > > can be revert if you want use hls_wrap.
> > >
> > >
> > > > There are many cases where the hls_wrap option remains critical so as
> > to
> > > > preserve the set of output file names without increments. This
> > > deprecation
> > > > breaks that.
> > > >
> > > > For now, this behavior can be worked around by switching to the
> segment
> > > > muxer and then setting the -segment_wrap option therein, but its' not
> > an
> > > > ideal solution.
> > > >
> > > > Kindly review this deprecation, with an appeal to revert the
> patchwork
> > > that
> > > > removed the hls_wrap option.
> > > >
> > > > Warm regards,
> > > >
> > > > Dennis.
> > >
> >
> > Steven,
> >
> > The option should've been left intact, *but* with a warning on its
> > implications on usage.
>
> Dennis,
>
> agreed with you, let me think about that.
>
> > Multiple ffmpeg flags have similar edge case impacts
> > when used improperly, eg -copyts and non monotonous timestamps, but
> they're
> > not deprecated; they're still in place.
> >
> > >
>
>
> Thanks
> Steven
>
>
>
Perfect, and thanks for considering a revert of this deprecated feature.
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Re: [FFmpeg-user] [FFmpeg-devel] FFmpeg's HLS muxer's deprecation of the hls_wrap option

[Steven Liu]

Dennis Mungai 于2024年1月20日 周六23:15写道：

> On Sat, 20 Jan 2024, 6:09 pm Steven Liu,  wrote:
>
> > Dennis Mungai 于2024年1月20日 周六21:02写道：
> >
> > > Hello,
> > >
> > > Is there a valid technical reason as to why FFmpeg's HLS muxer dropped
> > the
> > > hls_wrap option?
> >
> >
> > eg. three player playing the list and every fragment less or equal 1
> second
> > ,
> > 1. Player 1 from fragment1, fragment2
> > 2. Now Player 2 from fragment 1, fragment 2,
> > 3. Now Player 3 from fragment 1, Fragment2 and Player 1 playing fragment
> 3
> > 4. Network transport with player 3 get loss packet
> > 5. Player 1 playing new fragment 1, new fragment 2
> > 6. Player 2 playing fragment 3
> > 7. Player 3 blocking always because the fragments are flashing too fast.
> >
> > So the commit message said it is not friendly to downstream users.
> >
> > I have no more better way to fix it with ffmpeg, but I think that commit
> > can be revert if you want use hls_wrap.
> >
> >
> > > There are many cases where the hls_wrap option remains critical so as
> to
> > > preserve the set of output file names without increments. This
> > deprecation
> > > breaks that.
> > >
> > > For now, this behavior can be worked around by switching to the segment
> > > muxer and then setting the -segment_wrap option therein, but its' not
> an
> > > ideal solution.
> > >
> > > Kindly review this deprecation, with an appeal to revert the patchwork
> > that
> > > removed the hls_wrap option.
> > >
> > > Warm regards,
> > >
> > > Dennis.
> >
>
> Steven,
>
> The option should've been left intact, *but* with a warning on its
> implications on usage.

Dennis,

agreed with you, let me think about that.

> Multiple ffmpeg flags have similar edge case impacts
> when used improperly, eg -copyts and non monotonous timestamps, but they're
> not deprecated; they're still in place.
>
> >


Thanks
Steven

>
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Re: [FFmpeg-user] [FFmpeg-devel] FFmpeg's HLS muxer's deprecation of the hls_wrap option

[Dennis Mungai]

On Sat, 20 Jan 2024, 6:09 pm Steven Liu,  wrote:

> Dennis Mungai 于2024年1月20日 周六21:02写道：
>
> > Hello,
> >
> > Is there a valid technical reason as to why FFmpeg's HLS muxer dropped
> the
> > hls_wrap option?
>
>
> eg. three player playing the list and every fragment less or equal 1 second
> ,
> 1. Player 1 from fragment1, fragment2
> 2. Now Player 2 from fragment 1, fragment 2,
> 3. Now Player 3 from fragment 1, Fragment2 and Player 1 playing fragment 3
> 4. Network transport with player 3 get loss packet
> 5. Player 1 playing new fragment 1, new fragment 2
> 6. Player 2 playing fragment 3
> 7. Player 3 blocking always because the fragments are flashing too fast.
>
> So the commit message said it is not friendly to downstream users.
>
> I have no more better way to fix it with ffmpeg, but I think that commit
> can be revert if you want use hls_wrap.
>
>
> > There are many cases where the hls_wrap option remains critical so as to
> > preserve the set of output file names without increments. This
> deprecation
> > breaks that.
> >
> > For now, this behavior can be worked around by switching to the segment
> > muxer and then setting the -segment_wrap option therein, but its' not an
> > ideal solution.
> >
> > Kindly review this deprecation, with an appeal to revert the patchwork
> that
> > removed the hls_wrap option.
> >
> > Warm regards,
> >
> > Dennis.
>


Steven,

The option should've been left intact, *but* with a warning on its
implications on usage. Multiple ffmpeg flags have similar edge case impacts
when used improperly, eg -copyts and non monotonous timestamps, but they're
not deprecated; they're still in place.

>
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Re: [FFmpeg-user] [FFmpeg-devel] FFmpeg's HLS muxer's deprecation of the hls_wrap option

[Steven Liu]

Dennis Mungai 于2024年1月20日 周六21:02写道：

> Hello,
>
> Is there a valid technical reason as to why FFmpeg's HLS muxer dropped the
> hls_wrap option?


eg. three player playing the list and every fragment less or equal 1 second
,
1. Player 1 from fragment1, fragment2
2. Now Player 2 from fragment 1, fragment 2,
3. Now Player 3 from fragment 1, Fragment2 and Player 1 playing fragment 3
4. Network transport with player 3 get loss packet
5. Player 1 playing new fragment 1, new fragment 2
6. Player 2 playing fragment 3
7. Player 3 blocking always because the fragments are flashing too fast.

So the commit message said it is not friendly to downstream users.

I have no more better way to fix it with ffmpeg, but I think that commit
can be revert if you want use hls_wrap.


> There are many cases where the hls_wrap option remains critical so as to
> preserve the set of output file names without increments. This deprecation
> breaks that.
>
> For now, this behavior can be worked around by switching to the segment
> muxer and then setting the -segment_wrap option therein, but its' not an
> ideal solution.
>
> Kindly review this deprecation, with an appeal to revert the patchwork that
> removed the hls_wrap option.
>
> Warm regards,
>
> Dennis.
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Re: [FFmpeg-user] FFMPEG filter_complex overlays work well on windows but generates error on Linux and OSX

[Nicolas George]

Suraj Kadam (12023-12-22):
>for (let i = 0; i < pngPaths.length; i++) {

Hi. Nobody will try to understand your script in order to help you. You
need to simplify your command-line as much as possible while still
exposing the issue, and post it as is on the list along with the
complete console output.

Regards,

-- 
  Nicolas George


signature.asc
Description: PGP signature
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[FFmpeg-user] FFMPEG filter_complex overlays work well on windows but generates error on Linux and OSX

[Suraj Kadam]

So, I am applying multiple overlays on a video using FFMPEG, on windows it
always works fine for every size and number of overlays, but in case of MAC
or Linux, it generates error for reinitializing filters.

Let's say I have a video of 190MB, and about 182 overlay PNGs which will be
overlayed using filter_complex.

   for (let i = 0; i < pngPaths.length; i++) {
const start = Number(transcriptions[i].start).toFixed(2);
const end = Number(transcriptions[i].end).toFixed(2);
const overlayX = '0';
const overlayY = `0`;
filterComplex += `[${
  i + 1
}:v] overlay=${overlayX}:${overlayY}:enable='between(t,${start},${end})'`;
if (i < pngPaths.length - 1) {
  filterComplex += '[vout];[vout]';
}
  }

I am creating a filter_complex using above function.

And then running FFMPEG command like this:

  const outputPath =
  path.join(__dirname, '..') +
  `/projects/${project.originalVideoFile}-output.mp4`;
this.appGateway.sendProgress(JSON.stringify(userId), 'progress', 0);
return new Promise((resolve, reject) => {
  const ffmpegCommand = ffmpeg();

  // Add video input
  ffmpegCommand.input(videoPath);

  // Add PNG inputs
  for (const pngPath of pngPaths) {
ffmpegCommand.input(pngPath);
  }
  if (project?.user?.subscription?.plan?.id === 1) {
ffmpegCommand.input('watermark.png');
  }
  ffmpegCommand
.complexFilter(filterComplex)
.outputOptions('-c:v', 'libx264')
.output(outputPath)
.on('progress', (progress) => {
  this.appGateway.sendProgress(
JSON.stringify(userId),
'progress',
Math.round(progress.percent),
  );
})
.on('end', async () => {


"You can ignore the socket emmisions"

This works will on windows. But when processing on OSX or Linux, it
gives the following error:

Stream #95:0 (png) -> overlay (graph 0)
  Stream #96:0 (png) -> overlay (graph 0)
  Stream #97:0 (png) -> overlay (graph 0)
  Stream #98:0 (png) -> overlay (graph 0)
  Stream #99:0 (png) -> overlay (graph 0)
  Stream #100:0 (png) -> overlay (graph 0)
  Stream #101:0 (png) -> overlay (graph 0)
  Stream #102:0 (png) -> overlay (graph 0)
  Stream #103:0 (png) -> overlay (graph 0)
  Stream #104:0 (png) -> overlay (graph 0)
  Stream #105:0 (png) -> overlay (graph 0)
  Stream #106:0 (png) -> overlay (graph 0)
  Stream #107:0 (png) -> overlay (graph 0)
  Stream #108:0 (png) -> overlay (graph 0)
  Stream #109:0 (png) -> overlay (graph 0)
  Stream #110:0 (png) -> overlay (graph 0)
  Stream #111:0 (png) -> overlay (graph 0)
  Stream #112:0 (png) -> overlay (graph 0)
  Stream #113:0 (png) -> overlay (graph 0)
  Stream #114:0 (png) -> overlay (graph 0)
  Stream #115:0 (png) -> overlay (graph 0)
  Stream #116:0 (png) -> overlay (graph 0)
  Stream #117:0 (png) -> overlay (graph 0)
  Stream #118:0 (png) -> overlay (graph 0)
  Stream #119:0 (png) -> overlay (graph 0)
  Stream #120:0 (png) -> overlay (graph 0)
  Stream #121:0 (png) -> overlay (graph 0)
  Stream #122:0 (png) -> overlay (graph 0)
  Stream #123:0 (png) -> overlay (graph 0)
  Stream #124:0 (png) -> overlay (graph 0)
  Stream #125:0 (png) -> overlay (graph 0)
  Stream #126:0 (png) -> overlay (graph 0)
  Stream #127:0 (png) -> overlay (graph 0)
  Stream #128:0 (png) -> overlay (graph 0)
  Stream #129:0 (png) -> overlay (graph 0)
  Stream #130:0 (png) -> overlay (graph 0)
  Stream #131:0 (png) -> overlay (graph 0)
  Stream #132:0 (png) -> overlay (graph 0)
  Stream #133:0 (png) -> overlay (graph 0)
  Stream #134:0 (png) -> overlay (graph 0)
  Stream #135:0 (png) -> overlay (graph 0)
  Stream #136:0 (png) -> overlay (graph 0)
  Stream #137:0 (png) -> overlay (graph 0)
  Stream #138:0 (png) -> overlay (graph 0)
  Stream #139:0 (png) -> overlay (graph 0)
  Stream #140:0 (png) -> overlay (graph 0)
  Stream #141:0 (png) -> overlay (graph 0)
  Stream #142:0 (png) -> overlay (graph 0)
  Stream #143:0 (png) -> overlay (graph 0)
  Stream #144:0 (png) -> overlay (graph 0)
  Stream #145:0 (png) -> overlay (graph 0)
  Stream #146:0 (png) -> overlay (graph 0)
  Stream #147:0 (png) -> overlay (graph 0)
  Stream #148:0 (png) -> overlay (graph 0)
  Stream #149:0 (png) -> overlay (graph 0)
  Stream #150:0 (png) -> overlay (graph 0)
  Stream #151:0 (png) -> overlay (graph 0)
  Stream #152:0 (png) -> overlay (graph 0)
  Stream #153:0 (png) -> overlay (graph 0)
  Stream #154:0 (png) -> overlay (graph 0)
  Stream #155:0 (png) -> overlay (graph 0)
  Stream #156:0 (png) -> overlay (graph 0)
  Stream #157:0 (png) -> overlay (graph 0)
  Stream #158:0 (png) -> overlay (graph 0)
  Stream #159:0 (png) -> overlay (graph 0)
  Stream #160:0 (png) -> overlay (graph 0)
  Stream #161:0 (png) -> overlay (graph 0)
  Stream #162:0 (png) -> overlay (graph 0)
  Stream #163:0 (png) -> overlay (graph 0)
  Stream #164:0 (png) -> overlay (graph 0)
  Stream #165:0 (png) -> overlay (graph 0)
  Stream #166:0 (png) -> overlay (graph 0)
  Stream #167:0 (png) -> overlay (graph 0)
  Stream #168:0 (png) -> overlay (graph 0)
  Stream #169:0 (png) -> 

Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[Bouke / Videotoolshed]

> On 19 Dec 2023, at 17:24, Erik Dobberkau  wrote:
> 
>> 
>> @Shuto: I'm sure you've noticed already that you will want to adjust the
>> frame rate (option "-r"). But I'm not sure if that does also require an
>> adjustment of the -bufsize .
>> 
> 
> @Bouke: and it works without explicitly specifying a GOP size of 12 (for
> 50Hz  15 for 60-ish Hz) and a b-frame interval of 2 ?

Hi Eric,
Yup, this is the output of my (commercial) SpotXML app.
https://www.videotoolshed.com/product/spotxml/ 


Broadcasters over here have a Baton, an it eats the files.
(This particular one would be rejected cause having a non 5 sec multiple 
duration, and of course the volume change is very strange, resolution change 
probably not needed, I took the first stupid clip I found…)


Bouke

videotoolshed.com
Van Oldenbarneveltstraat 33
6512 AS Nijmegen, the Netherlands
+31 6 21817248
If you want to send me large files, please use:
https://videotoolshed.wetransfer.com/


> Erik
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[Erik Dobberkau]

> I understand that RDD-9 is necessary, but even after reading the article, I
> don't know where RDD-9 is.
>

RDD 9 is Sony's Registered Disclosure Document for the XDCAM file format
[1].
You can download the document from IEEExplore [2] if you have an account
(e.g. via a SMPTE membership, probably also via a ABU or JBA membership).

My latest (and probably outdated) information is that ffmpeg's MPEG2
encoder / MXF wrapper does not (yet, and for understandable reasons)
provide an option to reorder the B-frames as specifically required by RDD 9
(Closed GOP / Open GOP), and that's what a Sony hardware player marks as an
error.
The only (super easy to use) option that I know of to make the required
modifications is to run the file generated by ffmpeg through bmxtranswrap
[3][4].
The command is rather simple: bmxtranswrap -o /path/to/output_file.mxf -t
rdd9  /path/to/input_file_from_ffmpeg.mxf

(IIRC, I could write the bmx output file to XDCAM disc via a PDW U1/U2, and
play it back on a PDW F1600.)

But maybe there are other (free) alternatives available today that I'm not
aware of.

Erik

[1] https://www.smpte.org/standards/document-index/rdd#listing
[2] https://ieeexplore.ieee.org/document/7290306
[3] https://github.com/Phygon/bmxlib-bmx
[4] https://sourceforge.net/projects/bmxlib/files/
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[大石脩斗]

@Eril
Thank you for answering
I understand that RDD-9 is necessary, but even after reading the article, I
don't know where RDD-9 is.
I couldn't fix it myself.
Please tell me more details.



2023年12月20日(水) 1:16 Erik Dobberkau :

> >
> >  Trust me, this command (besides audio parsing, depending on sources) has
> > made thousands of broadcast compliant files over here, where XDCAM is
> also
> > mandatory.
> >
>
>  Does that mean the issue mentioned here:
> https://www.mail-archive.com/ffmpeg-user@ffmpeg.org/msg26313.html (and
> its follow-up messages) is not an issue anymore?
> I would expect that a piece of Sony XDCAM hardware expects a strictly RDD-9
> compliant MXF file (at least that's my experience with a PDW-F1600, never
> tried with a XDS-PD2000).
>
> @Shuto: I'm sure you've noticed already that you will want to adjust the
> frame rate (option "-r"). But I'm not sure if that does also require an
> adjustment of the -bufsize .
>
> Erik
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[大石脩斗]

@ Bouke
 Thank you for answering.
It didn't work.
I tried referring to your command, but the situation did not change.
Please let me know if there are any other improvements.
Also, it seems that the -map_channel option will be removed.
If you know an alternative pan filter command, please let me know.

command executed
%~dp0ffmpeg\ffmpeg.exe -hide_banner -threads 0 -i input/base.MXF -f lavfi
-i anullsrc=channel_layout=mono:sample_rate=48000 -map 0:0 -map 0:1 -map
0:1 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -c:a pcm_s24le
-ar 48000 -map_channel 0.1.0:0.0 -c:a pcm_s24le -ar 48000 -map_channel
0.1.0:0.1 -c:a pcm_s24le -map_channel 1.0.0:0.2 -c:a pcm_s24le -map_channel
1.0.0:0.3 -c:a pcm_s24le -map_channel 1.0.0:0.4 -c:a pcm_s24le -map_channel
1.0.0:0.5 -c:a pcm_s24le -map_channel 1.0.0:0.6 -c:a pcm_s24le -map_channel
1.0.0:0.7 -af volume=-11.4dB -video_format component -vf
setfield=tff,scale=1920x1080:flags=lanczos -r 29.97 -flags +ilme+ildct -top
1 -c:v mpeg2video -b:v 5k -maxrate 5k -minrate 5k -pix_fmt
yuv422p -profile:v 0 -level:v 2 -t 33 -g 15 -timecode 00:59:45:00 -metadata
creation_time=now -y output/output_test20.MXF


2023年12月20日(水) 0:33 Bouke / edit 'B :

> Top posting, as my mail client does now allow for decent parsing, and I’m
> fed up with that.
>
> My command to create ‘’proper” XDcam HD 50Mb, as required by the
> broadcasters here.
> Read between the filenames, it does work.
>
> /Applications/ffmpeg -hide_banner -threads 0 -i /Users/bouke/yadda.mov -f
> lavfi -i anullsrc=channel_layout=mono:sample_rate=48000 -map 0:0 -map 0:1
> -map 0:1 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -c:a
> pcm_s24le -ar 48000 -map_channel 0.1.0:0.1 -c:a pcm_s24le -ar 48000
> -map_channel 0.1.1:0.2 -c:a pcm_s24le -map_channel 1.0.0:0.3 -c:a pcm_s24le
> -map_channel 1.0.0:0.4 -c:a pcm_s24le -map_channel 1.0.0:0.5 -c:a pcm_s24le
> -map_channel 1.0.0:0.6 -c:a pcm_s24le -map_channel 1.0.0:0.7 -c:a pcm_s24le
> -map_channel 1.0.0:0.8 -af volume=-11.4dB -vf
> setfield=tff,scale=1920x1080:flags=lanczos -r 25 -flags +ilme+ildct -top 1
> -c:v mpeg2video -b:v 5k -maxrate 5k -bufsize 3835k -minrate 5k
> -pix_fmt yuv422p -profile:v 0 -level:v 2 -t 26 -timecode 00:00:00:00
> /Volumes/Data/magweg/Mars_Mars-Delight_26_1_19-12-2023_HD.mxf -y
>
>  Trust me, this command (besides audio parsing, depending on sources) has
> made thousands of broadcast compliant files over here, where XDCAM is also
> mandatory.
>
> Bouke
>
> > On 19 Dec 2023, at 06:56, 大石脩斗  wrote:
> >
> > I'm trying to use ffmpeg to generate an MXF file that can be played on an
> > XDCAM XDS-PD2000.
> > Please tell me what default processing is performed in ffmpeg when no
> > options are entered.
> >
> > [background]
> > Sony's check tool outputs an error saying "MXF metadata contains
> > non-standard values".
> > The data output by the editing machine is OK with the check tool,
> > The data generated with ffmpeg will be rejected by the check tool.
> > The metadata looks the same for both.
> > Please let me know if you are processing anything other than the meta
> > information that is being output
> >
> > 【Data output by editing machine」
> > aftertext.txt
> >
> > 【Data output by ffmpeg】
> > testffmpeg.txt
> > commondlie
> > ffmpeg.exe -i input\base.MXF -c:v mpeg2video -r 29.97 -pix_fmt yuv422p
> > -aspect 16:9 -flags +ildct+ilme -b:v 5000 -minrate 5000 -maxrate
> > 5000 -bufsize 17825792 -bf 2 -metadata creation_time=now
> -video_format
> > component -g 15 -timecode 00:59:45:00 -ac 1 -map 0:0 -map 0:1 -map 0:1
> -map
> > 0:1 -map 0:1 -map 0:1 -map 0:1 -map 0:1 -map 0:1 -c:a pcm_s24le -ar 48000
> > -f mxf -y output/output_test.mxf
> >
> > Shuto
> >
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[Nicolas Gaullier]

>> On 19 Dec 2023, at 17:16, Erik Dobberkau  wrote:
>> 
>>> 
>>> Trust me, this command (besides audio parsing, depending on sources) 
>>> has made thousands of broadcast compliant files over here, where 
>>> XDCAM is also mandatory.
>>> 
>> 
>> Does that mean the issue mentioned here:
>> https://www.mail-archive.com/ffmpeg-user@ffmpeg.org/msg26313.html 
>>  (and its 
>> follow-up messages) is not an issue anymore?
>
>No clue, only thing I know is that the stupid broadcasters over here accept 
>files created as I showed Bouke ___

Each broadcaster has its own requirements. They can be very vague or very 
specific. They can be justified, "fair" or they can be fictional.
They can be thoroughly checked by a QC system upstream, or only lately by 
simply checking if Image and Sound pops up.
There is definetely no universal "standard expectation".
Only thing is, when using native sony hardware, the file will be played out - 
or not, but this is a very rare use case.
Nowadays, most of the times, broadcasters ask for sony XDCAMHD but only require 
MPEG2-50 for real. But even so, you can face a "Baton dans les roues" (sorry 
for the poor French joke) : the enforced checks are often not related to the 
real world needs.
MXF interop should not be a big deal, but the xdcamhd essence can be, depending 
on your broadcaster expectations.
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[Bouke / edit 'B]

> On 19 Dec 2023, at 17:16, Erik Dobberkau  wrote:
> 
>> 
>> Trust me, this command (besides audio parsing, depending on sources) has
>> made thousands of broadcast compliant files over here, where XDCAM is also
>> mandatory.
>> 
> 
> Does that mean the issue mentioned here:
> https://www.mail-archive.com/ffmpeg-user@ffmpeg.org/msg26313.html 
>  (and
> its follow-up messages) is not an issue anymore?

No clue, only thing I know is that the stupid broadcasters over here accept 
files created as I showed
Bouke
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[Erik Dobberkau]

>
> @Shuto: I'm sure you've noticed already that you will want to adjust the
> frame rate (option "-r"). But I'm not sure if that does also require an
> adjustment of the -bufsize .
>

 @Bouke: and it works without explicitly specifying a GOP size of 12 (for
50Hz  15 for 60-ish Hz) and a b-frame interval of 2 ?

Erik
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[Erik Dobberkau]

>
>  Trust me, this command (besides audio parsing, depending on sources) has
> made thousands of broadcast compliant files over here, where XDCAM is also
> mandatory.
>

 Does that mean the issue mentioned here:
https://www.mail-archive.com/ffmpeg-user@ffmpeg.org/msg26313.html (and
its follow-up messages) is not an issue anymore?
I would expect that a piece of Sony XDCAM hardware expects a strictly RDD-9
compliant MXF file (at least that's my experience with a PDW-F1600, never
tried with a XDS-PD2000).

@Shuto: I'm sure you've noticed already that you will want to adjust the
frame rate (option "-r"). But I'm not sure if that does also require an
adjustment of the -bufsize .

Erik
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[Bouke / edit 'B]

Top posting, as my mail client does now allow for decent parsing, and I’m fed 
up with that.

My command to create ‘’proper” XDcam HD 50Mb, as required by the broadcasters 
here.
Read between the filenames, it does work.

/Applications/ffmpeg -hide_banner -threads 0 -i /Users/bouke/yadda.mov -f lavfi 
-i anullsrc=channel_layout=mono:sample_rate=48000 -map 0:0 -map 0:1 -map 0:1 
-map 1:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -c:a pcm_s24le -ar 48000 
-map_channel 0.1.0:0.1 -c:a pcm_s24le -ar 48000 -map_channel 0.1.1:0.2 -c:a 
pcm_s24le -map_channel 1.0.0:0.3 -c:a pcm_s24le -map_channel 1.0.0:0.4 -c:a 
pcm_s24le -map_channel 1.0.0:0.5 -c:a pcm_s24le -map_channel 1.0.0:0.6 -c:a 
pcm_s24le -map_channel 1.0.0:0.7 -c:a pcm_s24le -map_channel 1.0.0:0.8 -af 
volume=-11.4dB -vf setfield=tff,scale=1920x1080:flags=lanczos -r 25 -flags 
+ilme+ildct -top 1 -c:v mpeg2video -b:v 5k -maxrate 5k -bufsize 3835k 
-minrate 5k -pix_fmt yuv422p -profile:v 0 -level:v 2 -t 26 -timecode 
00:00:00:00 /Volumes/Data/magweg/Mars_Mars-Delight_26_1_19-12-2023_HD.mxf -y 

 Trust me, this command (besides audio parsing, depending on sources) has made 
thousands of broadcast compliant files over here, where XDCAM is also mandatory.

Bouke 

> On 19 Dec 2023, at 06:56, 大石脩斗  wrote:
> 
> I'm trying to use ffmpeg to generate an MXF file that can be played on an
> XDCAM XDS-PD2000.
> Please tell me what default processing is performed in ffmpeg when no
> options are entered.
> 
> [background]
> Sony's check tool outputs an error saying "MXF metadata contains
> non-standard values".
> The data output by the editing machine is OK with the check tool,
> The data generated with ffmpeg will be rejected by the check tool.
> The metadata looks the same for both.
> Please let me know if you are processing anything other than the meta
> information that is being output
> 
> 【Data output by editing machine」
> aftertext.txt
> 
> 【Data output by ffmpeg】
> testffmpeg.txt
> commondlie
> ffmpeg.exe -i input\base.MXF -c:v mpeg2video -r 29.97 -pix_fmt yuv422p
> -aspect 16:9 -flags +ildct+ilme -b:v 5000 -minrate 5000 -maxrate
> 5000 -bufsize 17825792 -bf 2 -metadata creation_time=now -video_format
> component -g 15 -timecode 00:59:45:00 -ac 1 -map 0:0 -map 0:1 -map 0:1 -map
> 0:1 -map 0:1 -map 0:1 -map 0:1 -map 0:1 -map 0:1 -c:a pcm_s24le -ar 48000
> -f mxf -y output/output_test.mxf
> 
> Shuto
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[FFmpeg-user] FFmpeg default processing(invisible processing)

[大石脩斗]

I'm trying to use ffmpeg to generate an MXF file that can be played on an
XDCAM XDS-PD2000.
Please tell me what default processing is performed in ffmpeg when no
options are entered.

[background]
Sony's check tool outputs an error saying "MXF metadata contains
non-standard values".
The data output by the editing machine is OK with the check tool,
The data generated with ffmpeg will be rejected by the check tool.
The metadata looks the same for both.
Please let me know if you are processing anything other than the meta
information that is being output

【Data output by editing machine」
aftertext.txt

【Data output by ffmpeg】
testffmpeg.txt
commondlie
ffmpeg.exe -i input\base.MXF -c:v mpeg2video -r 29.97 -pix_fmt yuv422p
-aspect 16:9 -flags +ildct+ilme -b:v 5000 -minrate 5000 -maxrate
5000 -bufsize 17825792 -bf 2 -metadata creation_time=now -video_format
component -g 15 -timecode 00:59:45:00 -ac 1 -map 0:0 -map 0:1 -map 0:1 -map
0:1 -map 0:1 -map 0:1 -map 0:1 -map 0:1 -map 0:1 -c:a pcm_s24le -ar 48000
-f mxf -y output/output_test.mxf

Shuto
General
Complete name: \output_test.mxf
Format   : MXF
Commercial name  : XDCAM HD422
Format version   : 1.3
Format profile   : OP-1a
Format settings  : Closed / Complete
File size: 236 MiB
Duration : 33 s 33 ms
Overall bit rate : 59.9 Mb/s
Frame rate   : 29.970 FPS
Encoded date : 2023-12-19 00:51:57.972
Writing application  : FFmpeg OP1a Muxer 60.18.100.0.0
Writing library  : Lavf (mingw32) 60.18.100.0.0

Video
ID   : 2
Format   : MPEG Video
Commercial name  : XDCAM HD422
Format version   : Version 2
Format profile   : 4:2:2@High
Format settings  : BVOP
Format settings, BVOP: Yes
Format settings, Matrix  : Default
Format settings, GOP : M=3, N=15
Format settings, picture structure   : Frame
Format settings, wrapping mode   : Frame
Codec ID : 0D01030102046001-0401020201040300
Duration : 33 s 33 ms
Bit rate mode: Constant
Bit rate : 50.0 Mb/s
Width: 1 920 pixels
Height   : 1 080 pixels
Display aspect ratio : 16:9
Frame rate   : 29.970 (3/1001) FPS
Standard : Component
Color space  : YUV
Chroma subsampling   : 4:2:2
Bit depth: 8 bits
Scan type: Interlaced
Scan order   : Top Field First
Compression mode : Lossy
Bits/(Pixel*Frame)   : 0.805
Time code of first frame : 00:59:45:00
Time code source : Group of pictures header
GOP, Open/Closed : Open
GOP, Open/Closed of first frame  : Closed
Stream size  : 197 MiB (83%)
Color range  : Limited
Color primaries  : BT.709
Transfer characteristics : BT.709
Matrix coefficients  : BT.709
Delay_SDTI   : 3588585

Audio #1
ID   : 3
Format   : PCM
Format settings  : Little
Format settings, wrapping mode   : Frame (AES)
Codec ID : 0D01030102060300
Duration : 33 s 33 ms
Bit rate mode: Constant
Bit rate : 1 152 kb/s
Channel(s)   : 1 channel
Sampling rate: 48.0 kHz
Frame rate   : 29.970 FPS (1601.6 SPF)
Bit depth: 24 bits
Stream size  : 4.54 MiB (2%)
Delay_SDTI   : 3588585
Locked   : Yes

Audio #2
ID   : 4
Format   : PCM
Format settings  : Little
Format settings, wrapping mode   : Frame (AES)
Codec ID : 0D01030102060300
Duration : 33 s 33 ms
Bit rate mode: Constant
Bit rate 

Re: [FFmpeg-user] ffmpeg and use of the ildct flag in progressive encoding

[Nik Armstrong]

On 2023-12-17 13:22, Bouke / Videotoolshed wrote:

On 17 Dec 2023, at 20:44, Nik Armstrong  wrote:

On 2023-12-16 15:01, Erik Dobberkau wrote:
The ildct flag appears only to be making the output file "think" it 
is
interlaced, when it is, in fact, progressive. It does not seem to be 
doing
anything to change the footage (quality-wise and it's still 
progressive).
Since someone other than myself set the flag on this command, I am 
trying
to figure out what effect, if any, taking it off will do to the 
output.
IIRC, it only has an effect when you are creating an interlaced 
output
using -top 1, (and from here on I‘m guessing) it might instruct the 
encoder
or an automatically inserted filter to apply the DCT in a different 
fashion

compared to a progressive input.
But again, that’s a guess which may (also) be outdated, since I last 
did

these kinds of transcodes quite a while ago.
Good luck.
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Thanks Erik. Much appreciated.


Mind you:
Sometimes it’s wanted / needed to have all files ’the same’.
If that is interlaced, it’s interlaced. Codec should say ‘interlaced’,
no matter the actual content.

For Avid media, that might (depending on the version of MC) be
important for a ‘fast import’, or omitting adapters you don’t want /
need.

In broadcast, over here interlaced is mandatory, even if the content
is progressive.

So no, it’s NOT complicated, and no, it’s NOT outdated.

Bouke

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Avid media: Interesting. That is what this command is being used for. 
However, when imported as interlaced (default by nature of the ildct 
flag) it causes problems if any time warps are applied (softens the 
image considerably). But when specifically instructed to import as 
progressive, it's not a problem.


Nik
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Re: [FFmpeg-user] ffmpeg and use of the ildct flag in progressive encoding

[Erik Dobberkau]

Sometimes it’s wanted / needed to have all files ’the same’.
> If that is interlaced, it’s interlaced. Codec should say ‘interlaced’, no
> matter the actual content.
>
> For Avid media, that might (depending on the version of MC) be important
> for a ‘fast import’, or omitting adapters you don’t want / need.
>
> In broadcast, over here interlaced is mandatory, even if the content is
> progressive.
>
> So no, it’s NOT complicated, and no, it’s NOT outdated.
>
> Bouke
>

I was only referring to applicability of my statement, not to the practice
as such — only taking the (likely) possibility into consideration that one
would have a different (i.e., better) approach to the required workflows
than I did about 10 years ago. If the solution is still applicable, I have
no reason nor intention to object or make a judgment.
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Re: [FFmpeg-user] ffmpeg and use of the ildct flag in progressive encoding

[Bouke / Videotoolshed]

> On 17 Dec 2023, at 20:44, Nik Armstrong  wrote:
> 
> On 2023-12-16 15:01, Erik Dobberkau wrote:
>> The ildct flag appears only to be making the output file "think" it is
>>> interlaced, when it is, in fact, progressive. It does not seem to be doing
>>> anything to change the footage (quality-wise and it's still progressive).
>>> Since someone other than myself set the flag on this command, I am trying
>>> to figure out what effect, if any, taking it off will do to the output.
>> IIRC, it only has an effect when you are creating an interlaced output
>> using -top 1, (and from here on I‘m guessing) it might instruct the encoder
>> or an automatically inserted filter to apply the DCT in a different fashion
>> compared to a progressive input.
>> But again, that’s a guess which may (also) be outdated, since I last did
>> these kinds of transcodes quite a while ago.
>> Good luck.
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>> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
> Thanks Erik. Much appreciated.

Mind you:
Sometimes it’s wanted / needed to have all files ’the same’.
If that is interlaced, it’s interlaced. Codec should say ‘interlaced’, no 
matter the actual content.

For Avid media, that might (depending on the version of MC) be important for a 
‘fast import’, or omitting adapters you don’t want / need.

In broadcast, over here interlaced is mandatory, even if the content is 
progressive.

So no, it’s NOT complicated, and no, it’s NOT outdated.

Bouke

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Re: [FFmpeg-user] ffmpeg and use of the ildct flag in progressive encoding

[Nik Armstrong]

On 2023-12-16 15:01, Erik Dobberkau wrote:

The ildct flag appears only to be making the output file "think" it is
interlaced, when it is, in fact, progressive. It does not seem to be 
doing
anything to change the footage (quality-wise and it's still 
progressive).
Since someone other than myself set the flag on this command, I am 
trying
to figure out what effect, if any, taking it off will do to the 
output.




IIRC, it only has an effect when you are creating an interlaced output
using -top 1, (and from here on I‘m guessing) it might instruct the 
encoder
or an automatically inserted filter to apply the DCT in a different 
fashion

compared to a progressive input.
But again, that’s a guess which may (also) be outdated, since I last 
did

these kinds of transcodes quite a while ago.
Good luck.
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Thanks Erik. Much appreciated.
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Re: [FFmpeg-user] ffmpeg and use of the ildct flag in progressive encoding

[Erik Dobberkau]

The ildct flag appears only to be making the output file "think" it is
> interlaced, when it is, in fact, progressive. It does not seem to be doing
> anything to change the footage (quality-wise and it's still progressive).
> Since someone other than myself set the flag on this command, I am trying
> to figure out what effect, if any, taking it off will do to the output.
>
>
IIRC, it only has an effect when you are creating an interlaced output
using -top 1, (and from here on I‘m guessing) it might instruct the encoder
or an automatically inserted filter to apply the DCT in a different fashion
compared to a progressive input.
But again, that’s a guess which may (also) be outdated, since I last did
these kinds of transcodes quite a while ago.
Good luck.
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